Archive for the 'Releases' Category



Version 1.8.5 is released

Version 1.8.5 is released, containing some bug fixes and incremental enhancements from previous releases. Please see the Download page as usual for more info.

 

PJSIP Version 1.8 Adds iPhone iOS4 Background Support

PJSIP version 1.8 has been released which adds iPhone/iPad iOS4 multitasking/background feature support into the library. For more info, please see the Release Notes from the Download page.

PJSIP version 1.7 adds iPhone and multipart support

PJSIP version 1.7 has been released, with two major features added, namely official PJSIP support for iOS (iPhone/iPad/iPod touch devices) and support for multipart message bodies. The iOS port added native CoreAudio audio device implementation usable for both Mac OS X and iOS, and it also utilizes the device’s built-in echo cancellation feature for more robust echo cancellation. The port also added built-in iLBC codec that comes with the device.

This version also fixed many bugs, part of which discovered during the last SIPIT 26 in Sweden.

Please see the usual download page for more info.

PJSIP version 1.6 is released

Many useful new features have been added on this release, as well as even more bug fixes. Please see the Release Notes for the complete info and get it from the usual Download page.

pjsip version 1.5.5 released: bug fixes only

A new version of pjsip has been released, it fixes 15 issues, no new features.

There’s something for everybody:

  • Platform fixes mostly for smartphones Windows Mobile and Symbian.
  • ICE (NAT Traversal) issues.

If you are not experiencing the issues listed, then there is no need to upgrade your pjsip.

You can download pjsip 1.5.5 right now.

pjsip version 1.5 is released with TLS Rewrite, TLS for Symbian, QoS, and MWI Support

Version 1.5 has just been released with the following features.

SSL/TLS Rewrite

A new secure socket abstraction is implemented in PJLIB. The API is implemented using the native CSecureSocket for Symbian platform and OpenSSL for other platforms. With this API, new type of implementations (such as native Windows SSPI)  could be written in the future.

The SIP TLS transport has been rewritten to make use of this secure socket API, while maintaining the existing SIP TLS transport API. The secure socket API will also make way for other SSL/TLS based transports in the future, such as TLS TURN client connection.

QoS Framework

All transports in the library (such as SIP UDP/TCP/TLS transports, UDP media transport, and STUN/TURN/ICE transports) have been equipped with QoS (Quality of Service) settings. The QoS framework abstracts QoS technologies such as the Type of Service/DiffServ Code Point (ToS/DSCP) fields, Wi-Fi Multimedia (WMM) priorities, and IEEE 802.1p tagging (via SO_PRIORITY) in a generic manner, while providing flexibility for applications to adjust the settings manually if wanted.

The QoS framework has been tested on Symbian, Windows Mobile 6, Linux, and MacOS X. Note that currently it is not available on Windows XP and later.

Please see the new QoS wiki page for more info.

Message Summary/Message Waiting Indication (MWI) Support

Added support for both subscription based MWI (RFC 3842) and unsolicited MWI that is used by a popular PBX. For more information please see ticket #982.

Presence Enhancements

Ticket #937 among other things implemented automatic buddy’s presence resubscription upon receiving several specific termination causes. Ticket #411 and #364 improved the PUBLISH request handling.

SIP INVITE/CANCEL Destination Fixes

Ticket #917 and #936 fixed the following problems:

  • CANCEL request may be sent to different server than the INVITE when DNS SRV is used
  • INVITE request retry because of 401/407 response may be sent to different server than the INVITE when DNS SRV is used
  • CANCEL request will be sent with UDP if the INVITE was sent with TCP because of 1300 bytes message size/MTU limit (it must be sent with the same transport)

Please get the new version from PJSIP download page as usual.

Version 1.4 is released with support for SIP Session Timer and Nokia VAS

“Good news, everyone!”

PJSIP version 1.4 has been released, with new features include SIP Session Timers (RFC 4028), support for VoIP Audio Services/VAS (including VAS-Direct) in Nokia FP1 handsets and newer, and initial porting to Symbian S60 5th Edition. Many bugs were fixed, as usual.

Please see PJSIP download page for more info.

Version 1.3 is released with support for ICE regular nomination

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 1.1 is released with support for Nokia native codecs and new audio device API

Good news, everyone!

Finally, after months of developments (read: delays!), version 1.1 is ready for your download. This release contains major feature enhancements, namely support for Nokia native codecs (we use code name APS-Direct for this feature) and a new Audio Device API.

APS-Direct is our codename for functionalities to use the hardware codecs that are supported by sound devices e.g. Nokia Audio Proxy Server (APS) and/or VoIP Audio Services (VAS) directly, bypassing media processing in PJMEDIA. The Nokia APS and VAS support codecs such as G.711 (PCMA and PCMU), G.729, iLBC, and AMR-NB, though the availability of these codecs may vary according to the handset types. There are significant benefits of using these codecs instead of software codecs (in PJMEDIA-CODEC), with the main benefits are performance (hardware vs software codecs, latency) and the given codec licenses/royalties. Due to these benefits, the ability to use these codecs in PJSIP applications is very desirable, hence the support.

This has been a major development in PJMEDIA, as traditionally PJMEDIA works with PCM (linear, L16) audio samples. With APS-Direct, audio frames from the sound device are in encoded format, so some components along the media path need to be updated to support encoded frame format. Understandably, some features cannot be used when encoded audio is active, for example the mixing feature of the conference bridge. Please see APS-Direct wiki for more information.

Half way during APS-Direct development, we discovered that the existing sound device abstraction API couldn’t cope with the new features, for example handling of encoded frames, setting the audio device routing, etc. We could of course patch it here and there, but we decided that creating a new one would be a much better alternative. So a new Audio Device API was developed.

For more information about this release, start from the PJSIP Download page. Enjoy!

We Made It: Release 1.0.1 Available

With the release of 1.0, we have reached a significant milestone. More than three years in development, 250,000+ lines of code, gruelling global interoperability testing on three continents, you can download pjsip 1.0 right now.

This would have never happened if it weren’t for all of you — more than 500 members of the pjsip community. A big thank you is in order.

We're talking through a megaphone

"Hello to All, we've reached 1.0! Thank you for all your support."

We’ve decided to draw a line on what we feel is a comfortable feature set, in particular for desktop client developers on Windows, Mac OS X, and Linux. Of course we can not fit it all, so keep those comments and requests coming, it matters to us.

For mobile platforms, consider the current feature set a technical preview. We know it’s not as easy to get started, and sometimes there are still issues. This is because mobile devices are really fragmented. You can have exactly the same device, but different firmware version can break applications.

"Mobile SIP SDK, anyone?"

"Mobile SIP SDK, anyone?"

Going forward we’ll strengthen and stabilize the mobile devices support, starting with Symbian S60 and Windows Mobile. We look forward to the Symbian Foundation open source version, more open always lead to more innovation.

I would also like to take this opportunity to formally introduce Teluu, as the company behind pjsip. It was started more than two years ago, alongside the pjsip open source project. Teluu is where we handle alternative licensing, direct developer-to-developer support, and the occasional customization work.

As always we welcome comments, feedback, both praise and criticism, and whatever else. You can leave a comment here, on the mailing list, or everywhere else on the web. As long as Google can find it, we should know about it 🙂