Archive for the 'Releases' Category



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 :)

Version 1.0-rc4 is released

Turns out that the previous release was not so much of a good news after all.

Several people had reported several build problems, especially on Mingw, and also on Windows Mobile, due to the recent addition of multicast constants in PJLIB socket abstraction.

We apologized for the problem (and quite humiliated too), hopefully we’ve learnt our lesson and this will not happen again in the future.  In the meantime, please find the update in the download page.

Version 1.0-rc3 is released

“Good news, everyone”.

PJSIP version 1.0-rc3 is ready. This release focuses on fixing bugs to make the libraries more stable for the upcoming 1.0 release. Some areas where bugs have been fixed include tone generator issues, SIP forking issues, and unexpected SIP message flow causes an assertion or crash.

Some enhancements have also been made on this release. Please see the ticket list in the Release Notes for the detailed list.

As usual, please go straight to the PJSIP download page for more info.

Version 1.0-rc2 is released

“Good news, everyone”.

PJSIP version 1.0-rc2 is ready. There’s not too many updates here (which is good, by the way), only 9 tickets on this release, but nevertheless some important bugs were fixed. So we’re inching closer towards the 1.0 release.

There’s not much else to discuss here, please go straight to the download page for the details.

Version 1.0-rc1 is released with new Python SIP, Nokia APS support, and IPP codecs

“Good news, everyone” [Professor Hubert J. Farnsworth, Futurama]

This is an interim release, intended to mark the end of features development in the trunk. From now on, it will be tests and bug fixes only, until we reach 1.0. This will be our first proper stable release, and it will be given a separate branch, to isolate it from bleeding edge developments in the trunk.

Because of that, there has been a bit of “pressure” to stuff in as much features as possible on this release, since this is the last change to include them in 1.0. Here are some of them:

  • Integration of Intel® IPP Codecs.  This brings us with bunch of new codecs into PJMEDIA, such as G.722.1, G.723.1, G.726, G.728, G.729A, AMR NB, and AMR WB. Basically the lot! For more info about this integration, please see here.
  • New Python API. We discussed this on this blog a month ago here, basically it’s a new Python API for PJSUA-LIB, it’s much easier to use, and it also has a more thorough documentation/tutorial. Please check that out.
  • Nokia APS Support. The Nokia Audio Proxy Server is a wrapper to Nokia S60 sound device, it has much lower latency than Symbian MMF API (the traditional sound device that we support), and it also opens up support for device’s native codecs such as AMR, G.729, and iLBC which we can use in the future.  Although this API has been deprecated by Nokia in FP2, still there are lots of S60 and FP1 devices out there, so this is worth supporting.
  • New Echo Suppressor. Good for mobile devices, we discussed this in this blog here.

And some more. For more information regarding this release, please visit the download page.

Enjoy.

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