Archive for the 'pjsip' Category

pjsip for BlackBerry PlayBook: Native SIP client may be closer than you think

Published 23 July 2011 Mobile VoIP , pjsip , Porting , SIP Client , smartphone 2 Comments
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One of the most frequently ask questions we got is: does pjsip support BlackBerry? Are there plans to do so?

We always answered no, because the current BlackBerry devices only supports Java and even then there seem to be lack of multimedia access (essential for audio/video capture and playback) and direct socket access (for media streaming).

Recently however, with the BlackBerry Playbook using a different operating system, they have also announced that native C/C++ development kit will be available later this year.

So, maybe it will be not long now until we can answer ‘Yes’. Well, at least it can run on the BlackBerry PlayBook, if not all their smartphones.

PJSIP 1.8.10 released with SIP outbound support

Published 7 December 2010 NAT traversal , pjsip , Releases Closed

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!

Native iPhone SIP Client Based on pjsip Available on App Store: Open Source and Not Tied to any Provider

Published 19 February 2009 pjsip , Porting 15 Comments
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pjsip on has been running on iPhone and iPod Touch for quite a while. Samuel Vinson (also responsible for making possible VoIP on Nintendo DS) was the first to announce a successful port to iPhone and iPod Touch even before the official SDK became available.

Siphon has already been available for developers and also on Cydia, an alternative distribution platform for iPhone applications. voiphone is another project starting up, based on sound device code from Siphon.

Now another milestone is reached, because an iPhone softphone called SipPhone on iPhone (how many phones can you have in a sentence!), has been released on the official App Store by VNet Corp of Shanghai. This means users unable or unwilling to install Cydia are also able to enjoy VoIP over Wi-fi with their favourite providers, instead of dictated by which client you use.

(For those reading on a computer with iTunes or on the iPhone itself here is the direct link to SipPhone on App Store.)

So how does it work? After downloading from App Store, following the installation instructions, I was able to add Teluu’s sipgate.co.uk account (look, No SIM!):

Main SIP account settings

Additional SIP account settings (optional)

I was then able to choose from my Contacts and make a call as normal. I didn’t do any extensive voice quality testing, just some quick calls. I will try to record some conversations to illustrate better the voice quality.

Another feature that needs pointing out is the ability to have multiple accounts. It was quite easy to toggle which account is active at any one time. The pjsip.org SIP domain uses OpenSER OpenSIPS, so I know this client is compatible with it.

Multiple accounts support for the iPhone SIP client

Multiple accounts support for the iPhone SIP client

The source of the application is available at their forum, it seems you can get it even if you are not a customer. This is beyond the requirements of the GPL, so nice touch on VNET Corp people.

I still haven’t been unable to compile it, so as can be seen I have a question pending there.

Overall of course the main issue of VoIP over wi-fi in iPhone remains: no background task. That means, unlike other mobile devices such as Nokia which uses Symbian, it cannot receive any calls while you are doing something else.

 

Let me know your comments if you have tried this iPhone SIP client.

Recognizing open source community contribution: Champion of the Month Award

Published 21 January 2009 Development Process , Open Source , pjsip 1 Comment
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At PJSIP we always feel we don’t express our gratitude well enough for all of your contributions to this project, be it comments, criticisms, bug reports, patches, and so on. Especially some people deserves our special thanks for their extra efforts in helping others, or for their perseverance in solving problems (read: bugs) when the going gets though.

Bug of the Month January 2009 Award

This award goes to the most most difficult, most elusive, or perhaps even most silly bug that was found this month. And the award goes to ….

bug #696

Bug #696

Bug #696. This had caused “RTCP clock skew detected” message to appear every now and then in the log and subsequently causing wrong RTT calculation.

What makes this bug a winner is that this has been so elusive that it had been living in the code for two and half years. We were aware of it but simply didn’t know how to fix it!

So our special thanks goes to Guido Fischer for solving this bug. You are smart. :)

Bug of the Year 2008 Award

The winner of Bug of The Year 2008 Award goes to …

bug #660

Bug #660

Bug #660, INVITE request is not retried with authentication, even when correct credential is available, if the 401/407 response is received after a 100 or provisional response (!).

This bug really deserves a special award. It is so trivial, and yet it had been living with us for as long as I can remember, it had passed so many tests by us and interop tests with so many products by so many people, and it even slip though undetected in two SIPit events. It simply cannot be more silly than this!

So we thank Bug #660 for truly shaming us all.

Champion of The Month Award

Champion of The Month earns our recognition for achieving expertise, participating in community activities, and sharing their knowledge with other community members in the PJSIP mailing list.  This month we would like to award this to ..

Gang Liu

Gang Liu

Gang Liu, for being actively helping other list members in the past few months, as well as contributing several patches. You have demonstrated your expertise with PJSIP and we as well as I’m sure others have found your contributions to be very valuable. You deserve this award!

Integrating yaSSL with pjsip: anyone interested?

Published 19 January 2009 Open Source , pjsip , Porting 2 Comments

Recently we’ve been in contact with the yaSSL open source SSL project, discussing possibility of supporting CyaSSL as well as the current OpenSSL.

yaSSL

yaSSL is a small footprint and portable SSL implementation

The small footprint and portability of CyaSSL seems a very good match, so anyone interested in kickstarting this? I guess this can be from both ends, so maybe a  yaSSL community member can try to see the TLS functionality in pjsip/pjmedia or from the pjsip side, looking into yaSSL stuff.

Let me know if you’re interested, and we can take it from there.

We Made It: Release 1.0.1 Available

Published 7 January 2009 pjsip , Releases 3 Comments
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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 :)

Presenting at CELF Embedded Linux Conference Europe 2008

Published 5 November 2008 Open Source , pjsip , VoIP Closed
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In the world of embedded Linux, one of the players is a group called Consumer Electronics Linux Forum (CELF). CELF was formed five years ago by the big manufacturers: Matsushita, Sony, Hitachi, NEC, Philips, Samsung, Sharp, and Toshiba.

One of the events they run is Embedded Linux Conference (ELC) series, which has been running the past three years and have two version, one in the US and one in Europe.

For the CELF ELC Europe 2008 we’re going to present a session on pjsip as an Open Source Compact SIP and Media Stack.

What’s the connection of pjsip to embedded Linux? Because pjsip is compact and lightweight it runs well in embedded Linux. We’re also going to talk about a recent case where Linux was used as the operating system for an IP Phone, and pjsip as the SIP stack, with pjmedia as the media stack.

Anyway, if you are anywhere near Ede, Netherlands on 6-7 November 2008, look us up!

UPDATE: The presentation is now available: pjsip: Open Source Compact SIP and Media Stack.

It’s Official: Mobile Goes Open Source

Published 23 October 2008 Mobile VoIP , Open Source , pjsip , Symbian 1 Comment
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Well, according to Symbian. I’ve been attending Smartphone Show 2008, because we have an open source Symbian SIP stack.

In the first two hours of the  I’ve been listening to Symbian itself, but also Motorola, Sony Ericcson, and Nokia.

I’ve lost count of how many times ‘open source’ is mentioned. Nigel Clifford, Symbian’s CEO, mentioned a figure of $300m in royalty revenues to be ‘given back’ to the ecosystem.

That’s a really clever way of putting it. He did admit some of them will be ‘pocketed’ instead of being used to ‘innovate’.

Which is another rather overused word during the whole show. One of the main argument for going open source is innovation over control.

Of course, one can always say Apple seemed to be able to out-innovate everyone in the mobile space, by having total and absolute control (Benoit Schillings, Qt CTO, alluded to a ‘dictator’ style of development).

So the strategy to beat Apple (and to a lesser extent Microsoft and Linux) is to club together in the Symbian Foundation, and using open source as the common denominator, a level playing field.

Number of devices already shipped, number of companies involved is frequently cited. It is hoped the ‘ecosystem’ will be large enough to eclipse (pun intended) other mobile platforms. The newly appointed Executive Director Lee Williams, also do not want to put a ’30% tax’ on the ecosystem.

It worked in the software tools space with Eclipse, but a mobile platform is a different beast altogether.

You can read up other thoughts on this matter from Dean Bubley and Simon Judge. Incidentally I can answer Simon’s wondering about UIQ: It’s officially dead. Patrik Olsson, VP and Head of Software at Sony Ericsson, said ‘you can get ready for Symbian Foundation-based Sony Ericsson phones now by using S60′.

Version 1.0-rc4 is released

Published 28 September 2008 pjsip , Releases 1 Comment

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

Published 25 September 2008 pjsip , Releases 3 Comments

“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.

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