VoIP-Forum

This week, the Asterisk Video Task Force has had it’s first workshop in Paris. (Nov 20-22, 2006)

First of all, we really have to send a big THANK YOU to Inria, the French research institute, that
hosted the event. They have been great hosts and provided us with both food, power and
network connectivity!
A special thank you to Philippe Sultan at Inria for organizing all of this.

To summarize:

1.4 testing

We started with testing the 1.4 beta. Some notes

• Direct RTP media setup did not work as expected. During the setup,
  the payload types was not copied between the call legs, resulting
  in bad calls with failing media. When answering, the capabilities
  of the callee is not used when sending 200 Ok to the caller. We opened a branch called “earlyrtpfix” to try to fix this. Almost there.
• IAX video is broken. Currently, the capabilities fails and video disappears
  somewhere in the call setup. Even with that fixed, there is a problem with
  not sending the frame sequence number across the bridge, which may
  result in bad resequencing on the other side. Out of order frames will be
  renumbered as they where in perfect order… IAX2 also seems to have
  problems with video frames larger than the MTU and video frames that
  are transported in multiple RTP payloads.
• We could set up calls between several phones. Ojo videophones,
  Aupix, Grandstream, Wooksung, Counterpath Eyebeam, Ekiga
  and Kapanga was tested as well as some unbranded softphones
  in development.
• We do have several problems in the RTP subsystem, that we
  need to look into.

Future things discussed

• Direct transcoding from alaw to ulaw. Module developed during the session by Morten.
• AMR codec support
• Integrations with 3G, support for 3G videocalls in libpri/chan_zap
• Exchange of more complete codec capabilities during call setup
  (branch videocaps)
• T.140 realtime text support
• Development of a contrib/videotoolbox for conversions, ivr video prompts etc
• Integration with remote transcoders
• Integration with OpenMCU videoconferencing
• Neil’s videoconference switch for app_conference
• H.323 video support
• Videolan integration for streaming video - VideoOnHold
• Next meeting in a sunny place
  After discussion, sunny is optional, no rain or snow is mandatory

This meeting was really productive and has moved Asterisk forward. Meeting for a few days really
helps, especially when you have a strong focus - like “video”.

/Olle Johansson

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