> I regret that I've only used MeetMe a few times, and only up to two users. > > Perhaps others that are using MeetMe could comment on the number of > concurrent conferences and total users they have asterisk running with. > The > specs of the systems involved would be most helpful. >
I have set up a conference with four people on a very low-end box. The voice quality was very good. All four were connected using VoIP. Three were using IAX2 clients, and one was using a SIP hardphone. I suspect the system could have easily scaled much further -- the CPU and memory usage were fairly low.
I've pushed a low-end system far enough to encounter problems. I set up what I thought was a prototype system on a box salvaged from the junkheap, and it immediately became so popular it got overloaded.
* 1, New Jersey:
P-II 350, 192 MB RAM, RedHat 8.0, * 1.0 stable
* 2, Jamaica:
P-III 800, 256 MB RAM, RedHat 8.0, * 1.0 stable7 party conference on * 1 in New Jersey. 3 extensions on * 2 via IAX trunk to * 1 (no jitter buffer). Four calls on * 1: 1 IAX to VoicePulse, 1 local SIP, 2 remote SIP on Xten softphones in Vermont and UK. All calls are GSM (including IAX trunk) except for 1 local SIP user who is ulaw.
7 users sounds OK *if* the network connection to Jamaica is in really good shape -- but audio quality on the Jamaica trunk goes downhill really fast as network quality deteriorates. It's *much* more sensitive to network quality than a regular call across the same trunk -- and it becomes more sensitive to network quality as the number of users in the conference increases. On one occasion, a 7-party conference ended because audio quality from Jamaica was unusable. I then immediately called one of the extensions in Jamaica, and audio quality was OK. CPU usage during the 7-party conference was usually 10 to 15%, with occasional peaks at 50%.
On a good day, the network path from NJ to Jamaica is about 170 msec with 15 msec jitter (sounds perfect); on a bad day, it's 10% packet loss and 500 msec jitter. Needless to say, the latter condition is always unusable. A one-to-one call is still usable with packet loss of 3 to 4% and 60 to 70 msec jitter (although it certainly has audible dropouts), but a 7-party conference sounds really dreadful.
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