Milestone release-1.1

Completed 8 years ago (03/19/09 14:46:56)

100%

Number of tickets:
closed:
34
active:
0
Total:
34
applications

1 / 1

common

3 / 3

pjlib

2 / 2

pjlib-util

1 / 1

pjmedia

12 / 12

pjmedia-audiodev

3 / 3

pjnath

2 / 2

pjsip

6 / 6

pjsua-lib

2 / 2

unit-tests

2 / 2

This release is the development in the trunk and contains both bug-fixes and features additions from the previous release-1.0.1. It may have non-backward compatible changes from the previous release. If you are maintaining a stable application based on the 1.0 series, you may want to have the release-1.0.2 instead.


Release Focus

This release contains major enhancements, with the introduction of Nokia APS-Direct and new Audio Device API.

Nokia APS-Direct

The APS-Direct is a feature to make use of the hardware codecs supported by Symbian's Audio Proxy Server (APS) API. Since audio frames coming from/into APS are in encoded format, we need to bypass most of the audio processing in PJMEDIA (since PJMEDIA works with PCM audio frames). Understandably many PJMEDIA features (e.g. conference bridge and many features associated with it such as WAV playback/recording) will not be available when APS-Direct is used.

See Using APS-Direct and VAS-Direct in PJMEDIA for more information.

New Audio Device API

With the introduction of APS-Direct, we discovered that the old sound device API couldn't provide us with the features that are required, so new audio device API has been developed.

See PJMEDIA Audio Device API for more information.

Automated-Tests

This release has been equipped with automated tests to maintain the quality of the code. Please see http://dash.pjsip.org for the result!


API Changes and Incompatibility Info

We would expect some changes in the sound device interface, with the addition of a new API to expose device capabilities (such as volume control and audio routing). See PJMEDIA Audio Device API for more information.


List of Enhancements

common

#732
Handle access point disconnection-reconnection on Symbian.
#738
APS-Direct: support for native codec in Nokia handsets

pjlib

No results

pjlib-util

No results

pjnath

No results

pjmedia-audiodev

No results

pjmedia

No results

pjsip

No results

pjsua-lib

No results

applications

No results

unit-tests

No results


List of Bugs

common

#739
Link error 'undefined reference to pjmedia_plc_*' when building for S60 3rd MR target.

pjlib

#703
Miscellaneous fixes for pjlib
#749
MacOS X endianness detection on universal build (thanks Ruud Klaver for the patch)

pjlib-util

#751
Incorrect table based CRC32 calculation on big endian platforms (thanks Ruud Klaver for the fix)

pjnath

#689
Deadlock caused by ICE media transport (thanks Alain Totouom for the report)
#742
Change in mapped/STUN IP address does not update ICE srflx candidate (thanks Alexei Kuznetsov for the report).

pjmedia-audiodev

No results

pjmedia

#680
Detection of handset's frame ptime in APS
#695
APS causes KERN-EXEC 0 error when a call has hung up (thanks Kevin Gong for the report)
#696
Wrong timestamp calculation causing RTCP clock skew (thanks Guido Fischer for the fix!)
#698
Segfault when calling pjmedia_transport_srtp_create() with pjmedia_srtp_setting set to NULL (thanks Ruud Klaver for the report).
#701
Possible heap corruption in pjmedia/pasound.c callback's thread (thanks Paulo Sousa for the detail report).
#710
Build fails when application uses pjmedia_snd_aps_activate_loudspeaker() (Thanks Quang Luong Thanh for the report).
#720
Miscellaneous fixes for pjmedia
#722
The case of no gap/burst may not be handled correctly in VoIP metrics calculations in RTCP-XR (thanks Johan Lantz for the report).
#728
Codec L16 open() doesn't initialize PLC & VAD settings (thanks Yann for the report).
#730
Wrong jitter buffer parameters set by the stream
#744
Jitter buffer does not discard duplicate frame if it is currently empty (thanks Mårten Wikström for the report)
#753
Wrong timestamp increment in pjmedia_clock (thanks Yann for the report)

pjsip

#692
Minor fixes for PJSIP
#713
Assertion in "../src/pjsip/sip_util.c:729: pjsip_process_route_set()" (thanks Ramesh D for the report)
#718
Crash when handling incoming request without rport (thanks Norma Steveley and Seth Hinze for the report)
#726
Bug in parsing tel: URI (thanks David Weidenkopf for the report)
#747
Bugs in parsing SIP torture messages (RFC 4475) (thanks Norma Steveley for the report)
#755
Problems with IPv6 SIP transport (thanks Cedric Levequ for the report)

pjsua-lib

#699
Auto-close sound device doesn't work when call disconnected without ever being confirmed/ringing (thanks Alexei Kuznetsov for the report).
#724
Miscellaneous fixes for pjsua-lib

applications

#716
Miscellaneous fixes for applications

unit-tests

#704
pjsip-test: "Bus error" on FreeBSD due to declaration of test data in read-only segment (thanks Michael Broughton for the report)
#707
Miscellaneous fixes for test module

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