Using p2p for conference call


#1

Hi all.

Is Asterisk possible to use p2p for conference call?
Because if it is possible, it would be helpful for save Asterisk’s server’s resources.

Sorry for my poop English.
Thank you for your help.


#2

Isnt a p2p call a call between two parties?

Why a conference call between two parties? But if it is a call between two parties then, yes you could use reinvite.

Maybe you have to explain a bit more what you want to achieve…

But you cant do p2p between more then 2 parties, at least not to my knowledge.


#3

Thank you for replying, meightee.

Sorry for my “poor” English.(Last time I wrote wrong spelling.)

Our customers want to send from one server a broadcast message to all currently registered sip clients.
I was thinking what would be the best way to do this?
I know that the “meet me” function could invite people into a conversation, but is this also possible with 100 active sip users at a time?
Next, to that I also know that there is a p2p function, would this be a better option for this?
Am I missing any possible functions here that Asterisk has for things like this?
What would be the best way to save server resources and to send a broadcast message to all the currently registered sip clients.


#4

Assuming you will use qualify option to determine if the sip devices are rechable and register, then use originate command to initiate the call, this will require posibly be done using AMI


#5

Hi, ambiorixg12.
Thank you for your reply.

I didn’t expect to use the “originate” function by AMI.
But if so, I couldn’t imagine how to send voice broadcasts to sip clients by using the “originate” function.
My goal is to send broadcasts to more than 100+ people at the same time.
Because, As far as I know, “originate” function is basically a one to one person call.
How would this be possible? Can you tell me more specific details about how to archive this?
Thank you for your sincere help.


#6

Originate can be called 100 times to originate 100 calls. It is the only practical way if people have to answer the call.

Page is an automated way of creating a of setting up lots of destinations in a muted conference, but command line length llimitations probably mean that you will still have to use originate to get the numbers,as well as to have a recording on the other side of the call. Page requires that you, somehow, set the phones to auto-answer. This can sometimes be done with a special SIP header, combined with disabling the security lockout on that header in each phone,beforehand.

All the above assumes your system has the physical capacity to make so many calls at once.


#7

Can your phones do Multicast? Polycom, Panasonic, and Yealink phones all support multicast paging.

You could Multicast a page out to any devices that have an active listener.


#8

Hi, david551, johnkiniston.

Thank you for sharing the best solution.
We try to use those functions(originate, page, auto-answer) for a test for sure.

Our customer is using iPads and Androids, and they made applications which are using sip libraries.
And for testing purposes, we are using pjsip.

I’m asking them about which sip libraries they are using now.
And I’ll post it as soon as possible.

Because I also want to know sip libraries are possible to “auto-answering”.

Thank you.