Published 7 December 2010
NAT traversal , pjsip , Releases
PJSIP 1.8.10 is released! As we’re currently busy with other development (namely, video for the upcoming 2.0; more on that later), we didn’t plan to put new features into this release indeed.
But still one new feature is worth mentioning. During our SIPit27 visit, we discovered that there are three proxy implementations that support SIP outbound extension (RFC 5626). We’ve always wanted to implement SIP outbound, because it’s very useful for NAT traversal, and the lack of support in the server side was the only thing that held us back. So this convinced us to write the extension on site, in time for successful participation in SIP outbound multiparty test on the event.
So that is the highlight of this release, namely SIP outbound support and one week worth of heavy QA at SIPit 27. Enjoy!
Published 3 July 2009
NAT traversal , Releases
Version 1.3 is out (finally!). No major feature was planned for this release, however there are few useful enhancements such as support for ICE regular nomination, SIP transport automatically switch to TCP when request is too large, and periodic 1 minute retransmission of provisional responses to prevent dialog from being destroyed by proxies, as well as many bug fixes.
Version 1.0.3 is also out, which contains bug fixes from both 1.2 and 1.3.
Get it while it’s hot from http://www.pjsip.org/download.htm
Version 1.2 has been released, among other things it contains:
- G.722.1(C) aka Siren7 and Siren14 codecs
- support for building Windows Mobile targets with Visual Studio 2005 (no more upgrading from embedded Visual C++)
- updated PJNATH for the latest STUN RFC and TURN draft.
Published 23 May 2007
NAT traversal , pjnath , Symbian
This is a major development since 0.5.10 series, with the new PJNATH library to support ICE, support for Symbian platform, and new third party libraries arrangement.
Please find the tarball and more info in the Download Page.
If you are a product manager wondering how to get into the VoIP market quickly before it moves to Telecom 6.0 or so, Jimmy Atkinson has helpfully provided a comprehensive list of 74 Open Source VoIP Apps & Resources. It’s all nicely categorized, with more than adequate descriptions.
You can just feel the raise you are going to get because it is just made it so easy to assemble and launch your product. Your boss will be amazed at your in-depth knowledge. Your development team will worship you as the Open Source God.
And yes, pjsip is listed as no. 31, by the way, in the category of SIP Protocol Stacks and Libraries. Actually to be a bit pedantic pjmedia should appear as well under RTP Protocol Stacks. And maybe pjnath, the new library for firewall traversal using ICE, listed under Development Stacks. But I digress. Go ahead, with 74 choices like that, is there any other reason NOT to go open source?