General Questions about Asterisk

I have no previous experience with Asterisk but I have very good knowledge and experience in Cisco Unified Communications. So, I know exactly, what is the meaning of VoIP, H.323, SIP, XMPP and etc.
I’m thinking to use Asterisk, but I really appreciate if someone can help me in the following points before reading and testing and at the end find that Asterisk is not applicable with what I’m looking for. The points are:
• Can I deploy Asterisk in Cloud and allow phones to register from Internet like homes?
• Is there an application that is similar to Cisco Jabber or Microsoft Lync (Skype for Business) where I can use it as:
o Softphone.
o Chat and instant messaging.
o Desktop sharing.
• What is the maximum number of phone registration that Asterisk can support?

Thanks in advance.


Most of the softphones support messaging(sip or xmpp), audio and video. Sharing a desktop is out of the scope of asterisk so you will need another software.

There is no limit on peer registration, actually there is no limit with nothing, that depends on your hardware specs.

Thank you too much for your reply.

Most of the softphones support messaging(sip or xmpp), audio and video. Sharing a desktop is out of the scope of asterisk so you will need another software.

Do you mean I have to use another application other than Asterisk or in addition to Asterisk. Like in Cisco: CUCM is the IP PBX and we need another server to work with CUCM to provide presence capabilities (IM and Presence Server)

So is there another application?

I was talking about the sharing Desktop option. Presence is supported by asterisk as well. You can take a look at the features that asterisk supports and then look at the sip client that you want to match your needs.

Basically in the world of open source you can create the things as you need it, quick example: you can create a WebRTC sip client(a softphone in your browser), most of the WebRTC-SIP APIs support presence and WebRTC support sharing desktop features. So it is up to you(your time and your money) how to sort all your needs.

Whilst you can run Asterisk in the cloud, it has a number of real time requirements which means you cannot just put it on any old virtual machine.

Ideally, you want maximum scheduling latencies to be less than 20ms (average ones a lot less) and system time to track real time to rather better than 20ms.

Thank you too much and I really appreciate your help. Do you have a link which list the features and the supported sip clients.

I will search about WebRTC and if I need help I will disturb you :slight_smile:

Thanks agian.

I’m not aware in Asterisk terms. But do you mean by scheduling latency the TTL between the nodes? in Cisco you have to make sure that you have enough bandwidth and speed between the nodes of the cluster.

I will go deeply in Asterisk in case I found it’s useful for my project.

Thanks :slight_smile:

The time between when a thread becomes runnable and it actually starts running. A thread is a single logically sequential stream of instruction processing on the CPU. E.g a thread that handles incoming media frames becomes runnable when the frame hits the ethernet card, and a thread that handles voice announcement becomes runnable when the next 20ms timing event happens.

Some VM Hosts may switch between different virtual machines, giving each one a few 100 milliseconds, which would be totally unusable for VoIP. Some will move processing between physical machines, taking 10s of seconds in the process, and that is totally unacceptable for VoIP.

Thank you too much for your reply. I will try to use a very powerful and supported VMs.


I downloaded AsteriskNow to do some testing. Regarding, the limitation of phone registration, can I use AsteriskNow with no limitation? based on the hardware only?

You can have as many extensions you want. Your hardware is the only limitation and how many concurrent calls can handle. Depending on your connection with the “outside” world (SIP or DAHDI trunk) the numbers of the calls vary. If you are using a DAHDI trunk this number drops significant because DAHDI calls are more cpu intensive than SIP.

Thank you for your support