2006 February

February 2006


This spring, I will be on tour quite a bit. In February, I am talking about Asterisk and SIP at the SIP International 2006 Conference in Paris. In March, I will be visiting Von 2006 and exhibiting with Digium in the Asterisk partner booth. In April, I am taking Asterisk to SIPit in Tokyo for one week of testing. And between those events, I have trainings in Europe and USA as well as customer meetings and Open Source and Linux user group meetings that invite me as a speaker.

In all of these trips, I will meet Asterisk users and developers in the Asterisk community. Open Source is really about people in addition to the software. By joining the Asterisk open source project, I now have friends all over the globe that I communicate with if not daily, then every week. Even in customer projects, community members help each other out, forming a large business network.

This is much more powerful than any commercial software I have worked with. Because an Open Source software is so much larger than the code itself. By forming a community as well as an business eco system, a piece of software grows extremely strong and is very hard to compete with. In fact, I can accept some problems in the software itself, because the power of the community makes me assured that I can get help almost around the clock.

Well, I have less license fees to pay by using Open Source. The problem is that airlines aren\’t yet Open Source. From my point of view, meeting all my friends and collegues in the community is worth it. See you!

One of the most common questions discussed within the Asterisk community is the upper limit for number of calls within a single Asterisk server. John Todd yesterday issued a challenge on the asterisk-dev mailing list. Quoting his mail:

The subject of load on a single chassis is still the most contentious issue to date. The Signate numbers of >5000 calls per chassis with RTP are impressive, and there are others who claim more vaguely of 1000, 2000, or more calls into a single P4 server (with or without media.) Others say that there are inherent limits in the Asterisk code which prevent more than ~500 calls from being processed with RTP at any one time. Opterons, FreeBSD, custom Linux loads, Solaris, and other operating systems or hardware have been offered as the magic bullets to increase call volumes. Who knows? (1) I will say that extraordinary claims demand extraordinary evidence, which has been pretty thin. I believe that most large call processing facilities still run on distributed systems of some type, as was described in the primary thread of this discussion on asterisk-users. (2)

I know that there are some projects towards testing Asterisk more rigorously to determine these numbers. However, I would suggest that the community at large could benefit from a more open examination of high-end system claims immediately than these (better) long-term tests which are progressing slowly (if at all.) Let’s just look at the “maximum” numbers. Running a big system? Selling a big system? Tell us about it, in detail. What are the limits that have been hit? Be specific. I keep seeing hand-waving, but no programmers have come forward to say “It won’t work because of the way X is implemented in the file blah.c or libFOO.”

To make a bad analogy: I don’t want to see the street rods; I just want to see the top-fuel, rocket-powered dragsters on the line.

Any takers? It sounds like Signate has a contender, but quite a few people have said that it’s impossible without serious modifications to the code. Others have claimed (publicly or privately) that they can match those numbers on different hardware.

Here are the criteria:

  • Any O/S
  • An unmodified version of Asterisk from SVN (or CVS)
    OR patches must be available for inspection, as per the GPL
    OR you must be a Digium license-holder (patches can be secret)
  • All calls are IAX2 or SIP (both in and out)
  • No transcoding of any type is required
  • All calls are G.711, 20ms OR 30ms packet size
  • All O/S documentation, kernel tricks, modules, hacks, patches, or configuration elements should be documented, but proprietary information need not be divulged if that is deemed “secret”
  • Testing method must be reasonably documented
  • Dialplans must be included
  • SIP.conf files must be included
  • All hardware must be fully described (part numbers required)

TEST #1:

  • All media must be handled by the server. This is for both legs of the call. The “canreinvite=no” for SIP and “notransfer=yes” in IAX2 must be set for all calls.

TEST #2:

  • Media may or may not be handled by the server. Native transfers should be allowed in both IAX2 and/or SIP.

(1) I have heard various people saying that it is “impossible” for Asterisk to handle a large number of calls due to architectural issues (no, it’s not just from the people that you’d “expect” to hear this from.) I’ve not been able to validate this one way or the other recently. I am interested to hear what the developer community has as a comment on this topic. I have an Empirix Hammer system at my company, but honestly I just don’t have the time to set it up to do testing due to day job time constraints…

(2) There are so many ways to spread calls across an Asterisk array it makes my head spin, but the question STILL comes down to “how many calls can a single chassis handle?” Even in a farm of servers, there has to be a numerator in that ratio.

JT

With the switch from CVS to subversion as a source code revision control system for Asterisk, a lot of new possibilities was created. One of them was branching. A branch is a copy of a repository where a developer or a team of developers can add new code without affecting the main version of the source. The extra benefit is that a branch maintains a relationship with the main source code. When new features are introduced or bugs fixed in the main source code, the changes are easily imported into the branch. When the development in the branch is done, the resulting changes can easily be merged into the main source code.

(more…)

Testing is important for every software project, commercial or free. We are now working on Asterisk 1.4 development and need your help testing new features that we are working on. A few weeks ago there was complaints about of the lack of testing new releases of Asterisk on this mailing list. Here is your big chance!

I have opened many subversion branches with different patches to chan_sip for testing. Go to the bug tracker, select the SIP category and you will find several interesting patches:

  • diversion: Support for the SIP Diversion: header
  • videosupport: Support for peer-configurable video support in SIP
  • jitterbuffer: Support for a jitterbuffer in chan_sip
  • strictrouting: Improved support for strict SIP routing
  • sipregister: Improved handling of register= statements
  • securertp: Secure RTP support, phase one

Some of these are work in progress, some are finished. All of them need your testing and input now, regardless if you are a user, system manager or a developer.

Help us make better software and more solid releases,
test patches and new code NOW!

Extra benefit: Testers get karma awards nowadays!

Meet you in the Asterisk bug tracker!

/Olle