The version 2.0 is still some way from the so called release quality, but for those who want to take a sneak preview on what’s going to be available, we just prepared PJSIP-2.0-Alpha for you. It’s available from the download page as usual. There is also a little Video Getting Started wiki to help you get things done quicker. Enjoy.
- PJSIP as the new SIP channel driver in Asterisk 12
- PJSIP Version 2.3 is Released with Video on iOS
- Native iPhone SIP Client Based on pjsip Available on App Store: Open Source and Not Tied to any Provider
- How to Use Your Nintendo DS as a Phone and Make Free Calls
- PJSIP Version 2.2 is Released with New API for C++, Java, and Python
- Command Line SIP Client
- Tiptel presents new IP phone series based on PJSIP 2.x
- PJNATH adds STUN, TURN, ICE to Asterisk for WebRTC support
- PJSIP version 2.2.1 is released
- Introducing pjnath - Open Source ICE, STUN, and TURN for NAT Traversal
- #1821: Remove unnecessary locking in pjsip transaction and add new API to create a group lock with handler in a single atomic instruction
- #1819: Use expiration field to indicate registration/unregistration in registration callback
- #1817: Automatically send BYE when sending CANCEL for INVITE is unsuccessful.
- #1814: Audio frame preview callbacks
- #1810: Adding CA path support into SSL socket
- #1818: Fixed destruction of locked mutex
- #1816: Restart media transport on following forked media
- #1815: Race condition of STUN transaction destruction (thanks to Itay for the report)
- #1813: Problem with media reinitialisation when using ICE
- #1811: contact_use_src_port and TLS server certificate verification issue (thanks to Viktor Krikun for the report)
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