<html><head><meta http-equiv="Content-Type" content="text/html; charset=us-ascii"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; line-break: after-white-space;" class=""><meta http-equiv="Content-Type" content="text/html; charset=us-ascii" class=""><div style="word-wrap: break-word; -webkit-nbsp-mode: space; line-break: after-white-space;" class=""><div class="">Hello,</div><div class=""><br class=""></div><div class="">There is a new release of SIP SIMPLE SDK and related command line clients.</div><div class=""><br class=""></div><div class="">sip-session script has been dramatically improved to the point of being usable as a fully fledge multiple account SIP client with end-to-end ZRTP for audio and OTR encryption for SIP MESSAGEs. </div><div class=""><br class=""></div><div class="">If one needs multiple SIP accounts for testing, they can now be created on the fly as well from within sip-session script or by using sip-settings.</div><div class=""><br class=""></div><div class="">To update or install the software go to</div><div class=""><br class=""></div><div class=""><a href="https://sipsimpleclient.org/installation/" class="">https://sipsimpleclient.org/installation/</a></div><div class=""><br class=""></div><div class="">Change logs</div><div class=""><br class=""></div>python-sipsimple (4.0.0)</div><div style="word-wrap: break-word; -webkit-nbsp-mode: space; line-break: after-white-space;" class=""><br class=""></div><div style="word-wrap: break-word; -webkit-nbsp-mode: space; line-break: after-white-space;" class=""><div style="word-wrap: break-word; -webkit-nbsp-mode: space; line-break: after-white-space;" class=""> * Migrated to python 3</div><div style="word-wrap: break-word; -webkit-nbsp-mode: space; line-break: after-white-space;" class=""> * Added migration filees fto or pjsip 2.10</div><div style="word-wrap: break-word; -webkit-nbsp-mode: space; line-break: after-white-space;" class=""> * Only publish Bonjour UDP transport if no other transport exists</div><div style="word-wrap: break-word; -webkit-nbsp-mode: space; line-break: after-white-space;" class=""> * Use TLS as default MSRP transport for Bonjour accounts</div><div style="word-wrap: break-word; -webkit-nbsp-mode: space; line-break: after-white-space;" class=""> * Update installation instructions</div><div style="word-wrap: break-word; -webkit-nbsp-mode: space; line-break: after-white-space;" class=""> * Show TLS error details in failure notification</div><div style="word-wrap: break-word; -webkit-nbsp-mode: space; line-break: after-white-space;" class=""> * Capture TLS certificates read errors</div><div style="word-wrap: break-word; -webkit-nbsp-mode: space; line-break: after-white-space;" class=""> * Do not verify TLS peer for Bonjour connections</div><div style="word-wrap: break-word; -webkit-nbsp-mode: space; line-break: after-white-space;" class=""> * Capture MSRP TLS verification error context</div><div style="word-wrap: break-word; -webkit-nbsp-mode: space; line-break: after-white-space;" class=""> * Load entire list of CAs</div><div style="word-wrap: break-word; -webkit-nbsp-mode: space; line-break: after-white-space;" class=""> * Post more XCAP notifications</div><div style="word-wrap: break-word; -webkit-nbsp-mode: space; line-break: after-white-space;" class=""> * Catch DNS resolver initializations exceptions</div><div style="word-wrap: break-word; -webkit-nbsp-mode: space; line-break: after-white-space;" class=""> * Added tls_name URI param to verify the peer inside PJSIP</div><div style="word-wrap: break-word; -webkit-nbsp-mode: space; line-break: after-white-space;" class=""> * Log which media stream has failed</div><div style="word-wrap: break-word; -webkit-nbsp-mode: space; line-break: after-white-space;" class=""> * Added tls_certificate ca_list properties for accounts</div><div style="word-wrap: break-word; -webkit-nbsp-mode: space; line-break: after-white-space;" class=""> * Fallback to tls general options if account tls settings are missing</div><div style="word-wrap: break-word; -webkit-nbsp-mode: space; line-break: after-white-space;" class=""> * Patch dns.query for versions that support setting the backend</div><div style="word-wrap: break-word; -webkit-nbsp-mode: space; line-break: after-white-space;" class=""><br class=""></div><div class="">sipclients (4.0.0) </div><div class=""><div class=""><br class=""></div><div class=""> * Migrate to Python3</div><div class=""> * Added outgoing file transfer to sip-session</div><div class=""> * Added video to sip-session</div><div class=""> * Added SIP message support for sip-session script</div><div class=""> * Added OTR encryption for SIP MESSAGE sessions to sip-session</div><div class=""> * Display GRUU URIs in sip-session and sip-register</div><div class=""> * Fixed logging when registration ended in sip-session script</div><div class=""> * Automatically changed default account after enrollment</div><div class=""> * Added python-requests dependency</div><div class=""> * Added <a href="http://sip2sip.info" class="">sip2sip.info</a> enrollment for sip-session script</div><div class=""> * Enable usage of multiple accounts with sip-session</div><div class=""> * Use environment set python2 path</div><div class=""> * Added ZRTP management for sip-session script</div><div class=""> * Added OTR management for sip-session chat stream</div><div class=""> * Preserve command history between restarts for sip-session script</div><div class=""> * Capture error when SIP session did not start yet</div><div class=""> * Moved spool directory under home user config folder</div><div class=""> * Added video support to sip-audio-session script</div><div class=""> * Print DNS lookup results for sip-message and sip-audio-session</div><div class=""> * Fixed printing XCAP document url</div><div class=""><div class=""> * Disable presence and xcap for sip-session</div><div class=""> * Disable presence and xcap for sip-audio-session</div><div class=""> * Disabled presence and xcap for sip-message</div><div class=""> * Enable sip trace for sip-message</div><div class=""> * Fixed publish presence script</div><div class=""> * Added sip and pjsip trace for sip-register</div><div class=""> * Fixed enabling tracing for audio session at start</div><div class=""> * Added possibility to start and end audio calls from external applications</div><div class=""> * Disable audio echo cancellation for arm7 architecture</div><div class=""> * Fixed starting publish presence application</div><div class=""> * Don't display key usage in batch mode for sip-audio-session</div><div class=""> * Fixed cancelling of incoming sessions for sip-audio-session</div><div class=""> * Added access list for auto answer for sip-audio-session</div><div class=""> * Unregister on exit for for sip-audio-session</div><div class=""><br class=""></div><div class="">Regards,</div><div class="">Adrian</div><div class=""><br class=""></div></div><div class=""><br class=""></div></div><div class=""><br class=""></div><div class=""><br class=""></div></div></body></html>