SIP SIMPLE Client SDK: Newshttp://projects.ag-projects.com/2016-03-09T15:09:10+01:00AG Projects
Redmine SIP SIMPLE client SDK 3.0.0http://projects.ag-projects.com/news/2812016-03-09T15:09:10+01:00Adrian Georgescu
<p>python-sipsimple (3.0.0) unstable; urgency=medium</p>
<ul>
<li>Added OTR encryption support in the chat stream</li>
<li>Use openfile where we need control over the file creation</li>
<li>Use defaultweakobjectmap defined in python-application</li>
<li>Simplified verifying the transferred file's hash</li>
<li>Added FileSelectorHash class to simplify code and improve readability</li>
<li>Fixed recovering session state in certain failure conditions</li>
<li>Do not wait for notifications if we couldn't notify transfer progress</li>
<li>Capture unhandled exceptions and log errors from new_from_sdp</li>
<li>Capture errors while parsing the file selector</li>
<li>Handle the MedisStreamDidNotInitialize notification when adding streams</li>
<li>Read code and reason from the notification when posting SIPSessionDidFail</li>
<li>Don't rely on the failure reason being set for failed transfers</li>
<li>Modified the SimplePayload and CPIMPayload to only work with bytes</li>
<li>Handle parsing errors for is-composing documents</li>
<li>Added package info module</li>
<li>Fixed MediaStreamBase not posting MediaStreamDidNotInitialize sometimes</li>
<li>Remove transfer_source from notifications</li>
<li>Set transfer origin to the remote identity if Referred-By is missing</li>
<li>Don't add a Referred-By header if it wasn't specified</li>
<li>Handle exception when closing a file that is being read in another thread</li>
<li>Prevent Session.transfer from being called while a transfer is in progress</li>
<li>Changed default transfer reject code from 486 (Busy) to 603 (Decline)</li>
<li>Protect SIPPublicationWillExpire from being called by an older publication</li>
<li>Handle race condition where state is SameState for initial PUBLISH</li>
<li>Remove bundled RFC/draft files</li>
<li>zrtp: prefer standard AES to Twofish cipher</li>
<li>Log Engine failures using application.log</li>
<li>Set locks to NULL after destroying them</li>
<li>pjsip: fix compilation warnings with recent versions of FFmpeg</li>
<li>pjsip: removed unused files</li>
<li>pjsip: update to revision 5249</li>
<li>Always build pjsip in non-debug mode</li>
<li>Avoid running timers if subscription dialog was destroyed</li>
<li>Suppress some compilation warnings</li>
<li>Avoid lockups on Engine shutdown</li>
<li>Post notification when SIPApplication gets a fatal error</li>
</ul>
<p><a class="external" href="http://sipsimpleclient.org">http://sipsimpleclient.org</a></p> SIP SIMPLE client SDK 2.5.0http://projects.ag-projects.com/news/2652015-12-03T15:06:32+01:00Adrian Georgescu
<p>New version with VP8 codec and several fixes</p>
<ul>
<li>Added VP8 video codec support</li>
<li>Added basic RTCP PLI support, used for requesting keyframes</li>
<li>Fixed crash when handling bogus Opus packets</li>
<li>Simplified registering audio codecs in the core</li>
<li>Handle socket errors when fetching XCAP documents</li>
<li>Fixed several compilation warnings</li>
<li>Added python-application as a build dependency</li>
<li>Fixed getting file offset on Windows</li>
</ul>
<p><a class="external" href="http://sipsimpleclient.org">http://sipsimpleclient.org</a></p> SIP SIMPLE client SDK 2.4.0http://projects.ag-projects.com/news/2372015-04-29T17:31:54+02:00Adrian Georgescu
<p>There is a new release of the SDK with MSRP file transfer pause/resume support and several bug fixes.</p>
<a name="Changelog"></a>
<h2 >Changelog<a href="#Changelog" class="wiki-anchor">¶</a></h2>
<p>python-sipsimple (2.4.0) unstable; urgency=medium</p>
<ul>
<li>Refactor file transfers and add resume support</li>
<li>Interrupt stream.initialize if session is cancelled</li>
<li>Add ISIPSimpleApplicationDataStorage interface</li>
<li>Adapt to API changes in MSRPlib</li>
<li>Simplified RTP streams initialization code</li>
</ul>
<p>python-msrplib (0.17.0) unstable; urgency=medium</p>
<ul>
<li>Remove file sending capability from MSRPSession</li>
<li>Fix sending response to unknown chunks</li>
<li>Fix building responses to non-SEND chunks</li>
<li>Refactor make_chunk into make_request and make_send_request</li>
<li>Removed leftovers from virtual chunking removal</li>
</ul> SIP SIMPLE client SDK 2.3.0http://projects.ag-projects.com/news/2212015-03-17T12:20:35+01:00Adrian Georgescu
<a name="python-sipsimple-230"></a>
<h2 >python-sipsimple (2.3.0)<a href="#python-sipsimple-230" class="wiki-anchor">¶</a></h2>
<ul>
<li>Added ZRTP support</li>
<li>Add setting for opportunistic SRTP encryption</li>
<li>Fix Opus SDP encoding</li>
<li>Reject video streams without a profile-level-id attribute</li>
<li>Renamed RTP stream related notifications</li>
<li>Renamed audio stream recording related notifications</li>
<li>Fix setting FFmpeg libraries path in some cases on OSX</li>
<li>Fix posting ICE state change notifications</li>
<li>Fix sending initial keyframes when ICE is used</li>
<li>Run _send_keyframes on the Twisted thread</li>
<li>Add ability to override the sender for chat messages</li>
<li>pjsip: fix rendering on OSX</li>
<li>pjsip: avoid crashes on OSX if video size changes</li>
<li>Work around issue with the RTP transport lock being hold for too long</li>
<li>Stop the VideoTransport in the device-io thread</li>
<li>Removed caching of statistics on the RTP transports when stopping</li>
<li>Fix draining the message queue in ChatStream</li>
<li>Post SIPApplicationWillStart before starting the core</li>
<li>Make SIPApplication start / stop consistent</li>
<li>Fix race conditions when handling SIPApplication.state</li>
<li>Avoid exceptions when un-pickling XCAP journal</li>
<li>Add chatroom_capabilities property to ChatStream</li>
</ul> ZRTP end-to-end audio and video encryptionhttp://projects.ag-projects.com/news/2172015-01-30T13:10:28+01:00Adrian Georgescu
<p>SIP SIMPLE Client SDK has now end-to-end encryption for audio and video calls using ZRTP protocol (RFC 6168 <a class="external" href="http://tools.ietf.org/html/rfc6189">http://tools.ietf.org/html/rfc6189</a>). The software is in trunk and can be fetched using darcs.</p> SIP SIMPLE client SDK version 2.0.0 releasedhttp://projects.ag-projects.com/news/2052014-11-21T13:31:56+01:00Adrian Georgescu
<a name="python-sipsimple-200-unstable-urgencymedium"></a>
<h2 >python-sipsimple (2.0.0) unstable; urgency=medium<a href="#python-sipsimple-200-unstable-urgencymedium" class="wiki-anchor">¶</a></h2>
<ul>
<li>Add video support (H264 codec)</li>
<li>Add support for handling initial INVITE requests without SDP</li>
<li>Restart TCP and TLS transports when network conditions change</li>
<li>Reuse disabled SDP streams</li>
<li>Fix parsing makefile options</li>
<li>Fix linking with OSX frameworks</li>
<li>Unregister and unpublish when network conditions change, before restarting</li>
<li>Leave symbols in even on release builds</li>
<li>Fix not ending streams in some cases when Session.remove_streams fails</li>
<li>Make stream initialization and ending consistent</li>
<li>Report stream failure in MediaStreamDidEnd</li>
<li>Add ExponentialTimer helper class to util</li>
<li>Improved handling failures when processing remote proposals</li>
<li>Reply with 488 if a remote SDP offer has deleted streams</li>
<li>Fix race conditions when handling remote proposals</li>
<li>Refresh SRTP crypto lines when updating local SDP</li>
<li>Simplify code for obtaining the path to the OSX SDK</li>
<li>Use 488 code when a proposal with streams fails</li>
<li>Use timezone aware timestamps for Session start_time and stop_time</li>
<li>Parse and generate bandwidth info attributes (b=) on the SDP</li>
<li>Include address information with the MSRPTransportTrace notification</li>
<li>Use SSLv23 method for TLSi (SSLv2 and SSLv3 are disabled)</li>
<li>Updated bundled PJSIP to revision 4961</li>
<li>Make sure a removed stream always has a connection line</li>
<li>Remove bandwidth attributes when disabling a stream</li>
<li>Reject incoming re-INVITE if we couldn't reply to it</li>
<li>Fix setting SDP connection line when accepting a proposal</li>
</ul>
<p>Note to SylkServer users: hold your update until a new release of SylkServer is made, adapting to some API changes in the SDK.</p>
<p>Note to Debian Wheezy users: You need to have the wheezy-backports repository enabled because we depend on backported libav. Also, make sure you have libxml2_2.8.0+dfsg1-7+wheezy1 installed, since the newest available version in wheezy is broken.</p>
<p><a class="external" href="http://sipsimpleclient.org/projects/sipsimpleclient/wiki/SipInstallation">http://sipsimpleclient.org/projects/sipsimpleclient/wiki/SipInstallation</a></p> Video Streams now in trunkhttp://projects.ag-projects.com/news/1932014-09-01T03:27:25+02:00Adrian Georgescu
<p>Current trunk has now video support! </p>
<ul>
<li>H.264 codec with full HD</li>
<li>ICE support</li>
<li>sRTP encryption</li>
</ul>
<p>Fetch it using darcs.</p> SIP SIMPLE client SDK version 1.4.2 releasedhttp://projects.ag-projects.com/news/1852014-07-29T14:19:24+02:00Adrian Georgescu
<a name="python-sipsimple-142"></a>
<h2 >python-sipsimple (1.4.2)<a href="#python-sipsimple-142" class="wiki-anchor">¶</a></h2>
<ul>
<li>Avoid recompiling the whole PJSIP every time the core is recompiled</li>
<li>Fix encoding when expanding user home path</li>
<li>Made the XMLDocument schema path configurable</li>
</ul>
<a name="sipclients-142"></a>
<h2 >sipclients (1.4.2)<a href="#sipclients-142" class="wiki-anchor">¶</a></h2>
<ul>
<li>Removed no longer needed future imports</li>
<li>Fixed handling html and multiline messages in chat</li>
<li>Fixed doubly defined notification handler</li>
</ul>
<p>The documentation for how to build the toolkit on Microsoft Windows has been updated.</p> SIP SIMPLE client SDK version 1.4.1 releasedhttp://projects.ag-projects.com/news/1772014-06-27T14:01:04+02:00Adrian Georgescu
<p>This release contains screen sharing using VNC over MSRP and several bug fixes and improvents.</p>
<a name="python-sipsimple-141"></a>
<h2 >python-sipsimple (1.4.1)<a href="#python-sipsimple-141" class="wiki-anchor">¶</a></h2>
<ul>
<li>Close external VNC viewer socket when the stream has ended</li>
<li>Don't try to set TLS options if there is no default account</li>
<li>Increased the connect timeout for external screen sharing handlers</li>
</ul>
<a name="python-sipsimple-140"></a>
<h2 >python-sipsimple (1.4.0)<a href="#python-sipsimple-140" class="wiki-anchor">¶</a></h2>
<ul>
<li>Add support for adding/removing multiple streams at once</li>
<li>Refactored SDP handling</li>
<li>Send re-INVITE after ICE negotiation is done</li>
<li>Refactored API for creating screen sharing streams and handlers</li>
<li>Enable RTP keep-alive using empty RTP packets</li>
<li>Disabled speex codec by default</li>
<li>Fixed race condition when saving ICE state</li>
<li>Made the VNC handler connection timeout a class attribute</li>
<li>Allow the default VNC server and viewer handlers to be configurable</li>
<li>Fix closing media transport to avoid leaking STUN sockets</li>
<li>Store our Python object in the user_data field of pjmedia_transport</li>
<li>Simplified srtp_active property</li>
<li>Make ice_active property dynamic</li>
<li>Make sure session state and proposed_streams are set to correct values<br /> when processing proposals</li>
<li>Use a shorter timeout for re-INVITEs that need to be answered without user<br /> interaction</li>
<li>Silence unused-function warning caused by python</li>
</ul> SIP SIMPLE client SDK version 1.3.0 releasedhttp://projects.ag-projects.com/news/1492014-04-11T13:33:13+02:00Adrian Georgescu
<p>python-sipsimple (1.3.0) unstable; urgency=medium</p>
<ul>
<li>Initialize core from main thread</li>
<li>Add AudioStream.recorder property and remove obsolete ones</li>
<li>Allow AudioStream.start_recording to be called early</li>
<li>Ensure MediaStreamDidEnd is always posted for MSRP streams</li>
<li>Fixed cancel_proposal when no streams were proposed</li>
<li>Fixed setting proposed streams on hold when holding during a re-INVITE</li>
<li>Fixed originator for SIPSessionProposalRejected</li>
<li>Fixed pickling for some core objects</li>
<li>Fixed compilation with latest Cython version</li>
<li>Fixed processing AudioPortDidChangeSlots if bridge was stopped</li>
<li>Avoid sending failure reports for MSRP keep-alive chunks</li>
<li>Avoid resetting greenlet when session is about to end or cancel a proposal</li>
<li>Removed unused tls_timeout configuration parameter</li>
<li>pjsip: don't compile libmilenage</li>
</ul>
<p><a class="external" href="http://sipsimpleclient.org/projects/sipsimpleclient/wiki/SipInstallation">http://sipsimpleclient.org/projects/sipsimpleclient/wiki/SipInstallation</a></p>