Bandwith for Conference Calls

Hi everyone!

My company is thinking about running Asterisk as a server for SIP based conference calls, because we had some problems in the past with Skype. The biggest problems seems to be that Skype simply adds bandwidths usage for each conference caller.

Now my question is, is Asterisk able to merge the streams of several callers into one stream to reduce the necessary bandwith? (Rather than sending one stream per caller to every caller in a conference.)

Many thanks in advance,
Oliver

Hi Oliver

Each user will use bandwidth. If more users are external than are internal use a external service such as conferencegenie or the like.

What you can do is set up an internal conference and then call the external conference and transfer that call into your internal conference, This way you will only have 1 call to outside .

Ian

Hi Ian,

thanks for the quick answer!

I know that every user will use bandwidth. This is not the actual problem…

Our situation is as follows. We have quite a few employees who call 3 to 4 people for a conference. The problem is that each call adds to our employees bandwidth usage as Skype simply creates streams from every caller to every caller.

As everyone should be able to speak with and to hear every caller in the conference, I thought there should be a possibility to have just one upload stream for each caller (rather than 3 or 4 as with Skype) to the server and one download stream (with all the other calls merged into one stream).

Hence, it would be necessary to merge audio streams on the server side…

Is that possible with Asterisk?

Oliver

Hi

[quote]I know that every user will use bandwidth. This is not the actual problem…
[/quote]

Yes it will be

As will any voip system. each call to a destination will use its own bandwidth

That is correct, I think the skype issue is that the caller who sets up the conference has added bandwith but this is peer to peer. Its a while since i studied the skype conference but its basicly teh same as any other conference. Im sure that every member doesnt get the stream form every other. But I may be wrong.

[quote]

Hence, it would be necessary to merge audio streams on the server side…

Is that possible with Asterisk? [/quote]

The audio is merged on server so 3 external users use the bandwidth of 3 calls.

I hope this is clearer

Ian