Sjur Usken

Views on new technologies and business opportunities from Sjur Usken

Callweaver, freeswitch and others are gaining speed!


The VoIP community is gaining speed, with good projects like Callweaver and Freeswitch! It is good to see that projects can evolve without the proprietary timers needed for the Digium version of Asterisk. It is fully understandable that Digium wants Asterisk to be dependent on their phone cards, but it hinders the VoIP development.

T.38 is important, even though the fax machine should have been replaced by a PKI system where you could send authenticated mails. It would also save the environment for prints…

The Callweaver (previous openpbx) can also be used as an session border controller. The SBC is really just a firewall, but with specific rules and algorithmes for VoIP. It is a product for the time beeing until the VoIP standards are fully complied by the different vendors. The SBC market will exist as long as all the vendors follows their “own” implementations of the standards.

Freeswitch is not based on Asterisk, but tried created better with the faults and drawbacks of Asterisk in mind. It can be used as a simple switching engine, a PBX, a media gateway or a media server to host IVR applications using simple scripts or XML to control the callflow. FreeSWITCH supports many advanced SIP features such as presence/BLF/SLA as well as TCP, TLS and sRTP. It also can be used as a transparent proxy with and without media in the path to act as a SBC (session border controller) and proxy T.38 and other end to end protocols.

I’m positive to the different flavors and distributions to Asterisk and hope they can learn from each other and bring the world forward!

Good job!

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4 responses to “Callweaver, freeswitch and others are gaining speed!

  1. leifmadsen June 18, 2008 at 5:06 am

    1) No hardware is required for timing (whether that hardware be Sangoma or Digium) — the ztdummy driver is all that is required.

    2) URL: http://svn.digium.com/view/asterisk?view=rev&rev=122928
    Log:
    Merge res_timing_pthread. This is a timing interface for Asterisk that does not require DAHDI. It’s called “pthread” because it uses a pthread API call in the timing thread for sleeping and ensuring we wake up at an appropriate time. I wasn’t sure what else to call it. 🙂

    The timing API requires a file descriptor that can be polled on. So, when you open a timer, this module creates a pipe and returns the read end of the pipe. There is a background thread that wakes up every 10ms and checks to see if any of the currently open timers need a ‘tick’ and writes to the appropriate pipe.

    3) … and russellb works for Digium none-the-less🙂

  2. sjur June 22, 2008 at 10:51 am

    Leif, excellent comment!

    The open-source PBX is definely a very strong alternative to Cisco, Avaya, and Microsoft OCS.

    I wish you all the luck with the next releases!

  3. anthm June 25, 2008 at 6:23 pm

    I think the title of this article helps suggest that FreeSWITCH is a derivative of Asterisk. Just to clarify, since FreeSWITCH is not mentioned anywhere in the actual article, FreeSWITCH is completely fresh code and has no connection to Asterisk.

    I have personally contributed code to the Asterisk project and spent many hours working on bugs etc. I can tell you I am certain that the dependency on the hardware came more from convenience than it did from a conspiracy theory.

    When I decided to write my own platform in 2006, I chose to make a timing interface such as this from the start anticipating the need to plug in alternate timing sources. I am glad to see that Asterisk is evolving in the right direction and I wish them luck.

  4. sjur June 25, 2008 at 10:26 pm

    Sorry about this! I have fixed it in the article as well! Great job!
    Are there any examples of FreeSwitch in REALLY large campuses with load-balancing and full redundant setup?

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