This latest release was supposed to be 0.7.1 a few months ago. That release was delayed, so more and more features got in. Therefore we have decided to call it 0.8.0. If you are still using 0.5.x series, we urge you download pjsip and upgrade pjsip now.
Here are excerpts from the release notes:
PRACK and UPDATE
Symbian support is getting more matured, with the implementation of Symbian sound device abstraction (ticket #2) and support for building the libraries as Dynamic Shared Object (DSO) files, which are needed for building developing for S60 3rd Edition using Code Warrior (ticket #354).
Updated STUN, TURN, and ICE
Custom SIP Presence Status Text
While previously PJSIP only supports basic online/offline status, now PJSIP supports specifying and receiving custom presence status text by implementing subset of RPID (ticket #336)
More robust NAT handling
For SIP, keep-alive mechanism has been implemented for UDP transport at PJSUA-LIB level (ticket #407), and both TCP and TLS transports at the transport level (ticket #95). Because of these the default registration interval is now extended to 5 minutes. The client registration session will also keep the transport open until it is destroyed, so that server can send SIP requests using this transport (mandatory for TLS, and could be useful for TCP) (ticket #390).
For SIP UDP transport, pjsua-lib by default (pjsua_acc_config.auto_update_nat setting) will monitor the STUN mapped address as reported by registrar. When it detects that the mapped SIP transport address has changed, it will unregister previous Contact, create a new Contact based on the new transport address, and restart the registration. This would happen automatically without application assistance (ticket #381).
For media, ICE transport will automatically change its transport address based on the address returned in the STUN keep-alive packets (ticket #372). Also pjsua-lib will now reports to application via a callback when ICE negotiation has failed (ticket #370).
More Robust SIP authentication
PJSIP now supports responding to authentication challenge for any realms, by specifying wildcard (“*”) as the realm in the credential (ticket #231). Although some have commented about security implications of this, a lot of people will find this feature to be very useful.
Basic support for 3GPP/IMS
Much improved audio latency on Windows
Audio latency on Windows (Win32) has been improved by several hundreds milliseconds. This should make the echo cancellation (AEC) works better too, so default EC tail length has been decreased from 800 ms to 200 ms.
Ticket #393 changed basic audio frame time, from 20 ms (hard coded as PTIME macro in pjsua_media.c) to 10 ms, and make this configurable. Default PortAudio sound driver backend was also made configurable, with the default is WMME (ticket #384). The default number of sound buffers (PJMEDIA_SOUND_BUFFER_COUNT) has been reduced from 16 to 6 (ticket #394). WMME audio latency buffering in PortAudio is now limited by 100 ms by default (ticket #395).