PJSIP version 1.7 adds iPhone and multipart support

Published 11 August 2010 Mobile VoIP , Releases , iPhone Leave a Comment
Tags:

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.

PJNATH ICE Heap Memory Usage Analyzed and Optimized

Published 5 June 2010 Embedded , NAT traversal , pjnath Leave a Comment
Tags:

A new article was posted in PJSIP wiki: PJNATH ICE Heap Usage Analysis and Optimization, that shows how to optimize ICE heap memory usage, from around 76 KB of peak heap usage per call (or 25 KB after the call settles down), down to just 21 KB of peak heap usage per call (or 15 KB after the call settles down). And this was with STUN, ICE, and TURN enabled.

PJSIP version 1.6 is released

Published 11 May 2010 Releases 4 Comments

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.

Migration of svn and Trac: expect disruption

Published 17 February 2010 Development Process Leave a Comment

We are currently migrating svn and trac to new hosting provider, so svn.pjsip.org and trac.pjsip.org will be disrupted for a little while.

Sorry for any inconvenience.

[Update: Everything should back to normal now. As a bonus, https connections are now properly certified without warnings! Yay!]

pjsip version 1.5.5 released: bug fixes only

Published 11 January 2010 Releases 4 Comments

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

Published 26 November 2009 Releases , VoIP 6 Comments

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

Published 19 August 2009 Mobile VoIP , Releases 1 Comment

“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

Published 3 July 2009 NAT traversal , Releases Leave a Comment

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 with support for Siren codecs

Published 20 May 2009 NAT traversal , Windows CE , Windows Mobile Leave a Comment

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.

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

Published 20 March 2009 Mobile VoIP , Releases , Symbian , smartphone 2 Comments

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!

Next Page »

Subscribe to blog updates

By email

RSS

RSS

Download pjsip

pjsip licensing

View Perry Ismangil's profile on LinkedIn

a

Clicky