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RELEASE NOTES
Version 0.5.6 2006/06/19
HIGHLIGHTS:
-
Complete rewrite of PJSUA-LIB (the highest layer of abstraction for PJSIP and
PJMEDIA) to make it a more "proper" high level API.
-
Initial support for RTEMS as target.
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Major modification in PJMEDIA.
-
Packet lost concealment (PLC) support with codec's built-in PLC or with
PJMEDIA's PLC add-in (e.g. for G.711 and GSM).
PJLIB:
New features:
Bug fixes:
PJLIB-UTIL:
-none
PJMEDIA
New features:
-
Basically there are pretty big changes in stream API.
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Handling of multiple frames in one RTP packet.
-
Split stream creation into two steps to allow customization in stream info.
-
Packet lost concealment (PLC) support with codec's built-in PLC or with
PJMEDIA add-in (e.g. for G.711 and GSM).
-
Jitter buffer correctly reports missing frames and this will activate the PLC.
-
Better handling of crappy sound device with DirectSound, and changed default
sound backend in Windows to use DirectSound.
-
Added device enumeration capability with DirectSound backend.
-
Added sound test in samples, to test sound device quality.
-
Introducing media transport class for stream, to allow non-UDP transport for
the media.
Bug fixes:
-
RTCP endianness, RTCP RR parsing, more tolerant to abnormal RTT value, and
tested with some hardware.
-
Reverse telephone-event payload type direction.
-
Jitter buffer is tolerant to RTP sequence number wrapping.
-
Timestamp in stream is not incremented when NULL frame is given.
-
Fixed bug in SDP negotiation, in the processing of answer from remote, when
dynamic payload type is used (it compared the payload type instead of the
name).
PJMEDIA-CODEC
New features:
-
The GSM codec was updated to patchlevel 12.
-
The speex codec was updated to 1.1.12.
PJSIP (Core, Ua, and SIMPLE)
New features:
-
Re-arrange transaction statefull stuffs so that it won't get linked when
transaction is not desired (to squeeze space).
-
The SIP status text in the response from remote is now kept in both transaction
and dialog, just in case remote sends customized status and application wants
to retrieve it.
Bug fixes:
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UAS dialog's remote.info_str was taken from local info.
-
Fixed bug in generating CANCEL request when the original INVITE request has
Route header.
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Fixed bug in client authentication, (1) tx_data is not invalidated, causing old
request to be sent, and (2) caching caused multiple identical headers to be
sent.
-
Increased default maximum SIP message size to 2000 (because 1500 is too small
if the message carries, for example, presence information in NOTIFY request).
PJSUA-LIB and PJSUA
New features:
-
PJSUA-LIB was completely rewritten! The API now should look more "proper" SIP
UA API than the old one, and this hopefully will make way for other SIP UA API
encapsulation, such as SIP ActiveX or SIP Python API.
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