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Re: [LUG] Well that was ... Intersting (SIP conference call)

 

On Sat, 16 Jan 2010, Dan Dart wrote:

Try having a small one before having a massive one... It will become
unmanagable above 4-5 people unless you are very, very disciplined.

True - 3 was bad enough but one was in another continent.

That's where VoIP has a slight disadvantage over the PSTN - lag/latency.

I've been involved in both unmanaged and managed conference calls while working for various companies - and it does get hard.

Things that go wrong - echo. Caused by bad/cheap speakerphones or using a PC speakers/microphone. Background noise - keyclicks especially, children howling for home workers... (Get a noise cancelling headset!) people falling asleep and snoring (ok, I've only heard this once ;-) and then there's people just talking over each other - this is especially frustrating with long-distance calls, espeically VoIP ones.

They can work, but they take discipline, or the use of a moderator who will only allow one person to speak at a time - and those sorts of facilities are expensive to run!

On the practical side - to run such a thing requires bandwidth - to make sums easy and build in overhead, assume 100Kb/sec each way, per call. So a home ADSL line is going to max-out at 4 external calls. A business one (830Kb/sec) at 8 calls. If you're lucky enough to have ADSL2+ with over 1Mb upstream, it's 10 calls at a push.

Don't forget it's also 50 packets per second each way. 10 calls, is 10 * 50 * 2 = 1000 packets per second that a router will have to manage - and those packets are 160 bytes each - not the sort of thing domestic routers are typically good at.

Or we can use a compressed codec - GSM or G726 is the best you'll get for free due to expired patents on the codecs, G.729 if you have a deskphone or pay for the professional version of X-Lite or Zoiper. The down-side of using compression is that the hosting server has to uncompress the incoming stream, mix the streams, then re-compress the outgoing ones that need to be compressed. All in real-time, so that imposes processing load on the server - not to mention the need for paid-for licenses if using G729 (if you believe in software patents that is)

Skype uses some sort of compression and interstingly enough when you buy the licensed version for Asterisk is includes a G729 codec license - possibly implying that Skype uses G729 internally... Can skype support conference calls? If so, the host is doing the uncompressing, mixing and re-compressing of the data... Quality suffers.

So, technology aside, if you can find somewhere in the UK with lots and lots of free bandwidth (10 people dialling in is the best part of 1Mb/sec each way all the time), then go for it...

Oh - and as an excercise, work out the data usage for 24 hours at 100Mb/sec each way and see if your broadband package will cope. Just in-case you leave it running...

Gordon

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