Hi Asterisk and VOIP Expert,
I want to setup a new VOIP business using asterisk. The requirement is as below:
Asterisk Server 1----------|
| My central
Asterisk Server 2----------| ----SIP Call Forward---> [Asterisk Server]--Trunk-->Call route to any Number
| Billing and Charging
Asterisk Server 3----------| +
CRM
can any one guide how to use my central asterisk server from other asterisk server and route to other part of the word. I mean call establish to any phone. For the call establish to out world..what SIP need to use? Or i need to install digisum cards on my central asterisk server or what?
Target is 20,000 Concurrent call. How concurrent call Asterisk 11.4 can handle?
Please cooperate for any information, or any link or diagram.
Usually you want to forward your outside calls to a VoIP provider (using SIP), so them can be routed to any number worldwide.
Take a look at http://www.voip-info.org/wiki/view/VOIP+Service+Providers
Digium cards, as interface cards of any other manufacturer (Sangoma, for example), could be used to connect your asterisk box to a digital or analog interface of a telecom. Which allows you to receive and send calls over.
First way is less expensive, but much unreliable. Second gives you solid business solution, but charges rise significantly when you start making long-distance calls.
You need to setup extension for each of your servers on central server.
After that you have use extension username/secret for trunk settings on other servers.
http://www.voip-info.org/wiki/view/Asterisk+Connect+2+servers
Unfortunately asterisk not able handle 20000 calls. Hi load servers(more then 200 calls) have be setup by experts, if more then 400 call - using other technologies like opensips(ser,kamailio)+rtprpoxy.
About asterisk hiload: fine tuned asterisk can handle 400-700 calls with media
Without media(just setups,media go directly) asterisk can handle 20-30cps on hi-end servers like 3.2Ghz 8-core xeons:
-- upto 4500 calls with ASR 50% and ACD 5minutes or less if ASR/ACD go down)
-- upto 1250 calls with typical ASR 25% and ACD 2 minutes.
-- only 20-30 calls when no success,so such setup can't be considered production worth.
Note, setup without media WILL have issues if both legs not trusted.
Related
I want to setup a personal videoconferencing service for my family, friends and myself. The main problem I have with current options is that they are either closed-source and centralized (GG hangouts, skype) or open-source but not working in corporate environment or in hotels (due to strict firewalling rules and the "Skype is going through, if you want VOIP use that" kind of netadmin reaction).
I have two solutions then. Either setup a STUN/TURN relay server and use XMPP and SIP as I used to, but that would require my friends to setup that too. Or setup a whole VOIP server. 2 solutions come to mind: SIP and XMPP. Though to my knowledge, each of them ultimately uses the (S)RTP/RTCP protocol.
And that's the problem. Out of the specific signaling part used by the two of them, I really can't figure out the difference between them, their typical use case.
I think you're right in that as far as setting up a video conferencing system XMPP and SIP are equivalent. They both are signalling only protocols and the media sessions they set up typically use RTP (although they can both be used to set up any kind of session you want but RTP is the norm).
The biggest problem is also going to be the one you mention about getting video streams out of a corporate firewall. Skype overcomes this obstacle by sending it's media over an SSL connection and is thus able to get through firewalls. Theoretically you could do the same with RTP and in the past I once used openvpn connections with a SIP client to test some audio calls. My experience wasn't great as the audio was very choppy, assumedly as a result of all the extra packaging that is required to get the high volume of small audio packets from one end to the other. That was nearly a decade ago though so perhaps with the better CPU and bandwidth resources available now it would work better.
Personally I think I'd stick with Skype as it's going to be a big hassle to set up your own system. If you were to go ahead with your own the first option I would try would be Asterisk combined with openvpn so that if the clients were behind a firewall or had NAT issues they could connect over it.
I setup an Active/Passive cluster with Pacemaker/Corosync/DRBD. I wanted to make an Asterisk server HA. The solution works perfectly but when the service fails on one server and starts on another all registered SIP clients with the active server will be lost. And the passive server show nothing in the output of:
sip show peers
Until clients make a call or register again. One solution is to set the Registration rate on clients to 1 Min or so. Are there other options? For example integrating Asterisk with a DBMS helps to save this kind of state in a DB??
First of all doing clusters by non-expert is bad idea.
You can use realtime sip architecture, it save state in database. Complexity - average. Note, "sip show peers" for realtime also show nothing.
You can use memory duplicating cluster(some solution for xen exists) which will copy memory state from one server to other. Complexity - very complex.
I have a client company with a simple web application (Python Flask) and I need to add a phone notification functionality to it.
The main requirement is that the app should call users, play a certain sound file and accept some tone input ("Hello! This is an automated message from your WebApp account. You have a meeting with $John today at $5pm. Please press 1 to confirm").
The other requirement is that the solution should be relatively cheap and fast to market.
I have done some research already and it seems that there are a few consequent steps to achieve that:
Set up an Asterisk or a FreeSwitch server;
Set up a SIP account;
Write some business logic for the Asterisk server which allows to make calls and play sounds via a SIP account;
Write an API at the Asterisk server and expose it to the Python Flask web app.
Do I miss something here? Can any of the steps be omitted anyhow? Can I do it simpler?
the fastest way to get it working is to use one of the cloud voice services with speech synthesiser. Here's a short list to check out:
Twilio
Tropo
Plivo
Here I listed some details.
Those services charge you per minute, plus you may have to pay some monthly fee.
If you want to run an independent and standalone service, I would recommend FreeSWITCH instead of Asterisk. It's got reach integration possibilities and API. You will need to read the FreeSWITCH book in order to understand how it works and how to build your service.
I agree with Stanislav Sinyagin on the cloud based solutions, but I would add one more, Voxeo Prophecy. Tropo is from Voxeo, but they have offered Prophecy as a solution for a lot longer and it supports the open standards CCXML and VoiceXML. The advantage of CCXML for outbound notification applications is you have a lot more control of the notification process.
The Prophecy platform has excellent call progress analysis (CPA) which will allow you to determine whether a machine or a human answered and handle the call accordingly. For example, it does not make sense to ask a machine to "...press one to confirm". Instead you may want to leave a message that provides a call back number for the user to confirm with after they have listened to the voice message. The CPA can be used to leave a message on a machine at the correct time (when the greeting message has stopped) so that you do not get clipped messages in the voice mail. CPA will also allow you to provide detailed reports on who was notified and for those that did not it can tell you whether it was a bad number (received a SIT tone), a modem or fax answered, or ring-no-answer (pretty rare these days). These type of details can factor into your retry process for failed notifications.
The other advantage to using Prophecy and open standards is your application will be portable to other IVR systems that are VoiceXML/CCXML compatible if you ever want to migrate. Tropo, Twilio, and Plivo all use proprietary API's which does not allow you to move your applications to other services. Prophecy is also available as a software solution so that if you want to take it out of the cloud you can run it on premise. You can get a two port version for free to try it out.
There is excellent documentation on developing outbound notification systems on Voxeo's developer site. Take a look at the CCXML documentation in section F on Outbound Dialing.
Not sure which development languages you are familiar with, but if you are used to ASP.NET MVC there is an open source project called VoiceModel that makes it easier to develop VoiceXML applications. The other advantage of VoiceModel is that you develop your application once and it will run on any VoiceXML compatible platform and Tropo. They are currently working on adding outbound notification support in this project that will work for both Tropo and VoiceXML.
Third party solutions listed are your easy choice. Running your own asterisk is also suitable for what you want to do, but i think for only this much it would be overkill, from an operational perspective.
In asterisk, you can originate a call that has the 2 variables you need with an (basic-authenticated) HTTP request. You will also need some settings and a tiny dialplan. Setting up the SIP account is easier or more difficult, depending on the documentation from the provider. Most of them have detailed documentation for configuring asterisk (not so much so for freeswitch). Keeping the damn thing alive is what's gonna get to you :)
I am building a distributed system that consists of potentially millions of clients which all need to keep an open (preferrably HTTP) connection to wait for a command from the server (which is running somewhere else). The load of messages / commmands will not be very high, maybe one message / sec / 1000 clients which means it would be 1000 msg/sec # 1 million clients. => it's basically about the concurrent connections.
The requirements are simple too. One way messaging (server->client), only 1 client per "channel".
I am pretty open in terms of technology (xmpp / websockets / comet / ...). I am using Google App Engine as server, but their "channels" won't work for me unfortunately (too low quotas and no Java client). XMPP was an option but is quite expensive. So far I was using URL Fetch & pubnub, but they just started charging for connections (big time).
So:
Does anyone know of a service out there that can do that for me in an affordable way? Most I have found restrict or heavily charge for connections.
Any experience with implementing such a server yourself? I have actually done that already and it works pretty well (based on Tomcat & NIO) but I haven't had the time yet to actually set up a large load test environment (partially because this is still a fallback solution, I'd prefer a battle hardened msg server). Any experience to how many users you get per GB? Any hard limits?
My architecture also allows to fragment the msg servers, but I'd like to maximize the concurrent connections because the msg processing CPU overhead is minimal.
I have meanwhile implemented my own message server using netty.io. Netty makes use of Java NIO and scales extremely well. For idle connections I get a memory footprint of 500 bytes per connection. I am doing only very simple message forwarding (no caching, storage or other fancy stuff) but with that am easily getting 1000 - 1500 msg / sec (each half a KB) on the small amazon instance (1ECU / 1.6GB).
Otherwise if you are looking for a (paid) service then I can recommend spire.io (they do not charge for connections but have a higher price per message) or pubnub (they do charge for connections but are cheaper per message).
You have to look more in architecture of making such environment.
First of all, if you will write sockets management by yourself, then don't use Thread per Client Socket. Use Asynchronous methods for receiving and sending data.
WebSockets might be too heavy if your messages are small. Because it implements framing, which has to be applied to each message for each socket individually (caching can be used for different versions of WebSockets protocols), that makes them slower to process both directions: for receive and for send, especially because of data masking.
It is possible to create millions of sockets, but only most advanced technologies are capable to do so. Erlang is able to handle millions connections, and is pretty scalable.
If you would like to have millions of connections using other higher level technologies, then you need to think about clustering of what you are trying to accomplish.
For example using gateway server that will keep track of all processing servers. And have data of them (IP, ports, load (if it will be one internal network, firewalling and port forwarding might be handy here).
Client software connects to that gateway server, gateway server checks the least loaded server and sends ip and port to client. Client creates connection directly to working server using provided address.
That way you will have gateway which as well can handle authorization, and wont hold connections for long, so one of them might be enough. And many workers that are doing publishing of data and keeping connections.
This is very related to your needs, and might not be suitable for your solutions.
I have a chat application (xmpp / muc) that is going to be served by apache (we might change to nginx later but right now it's not easily done). If a user is in 2 rooms, he'll have between 2 and 4 active connections to the server (long-polling connections), so if we have 200 users per room and we have 5 rooms, what should the ServerLimit, MaxClients be set to? For example, these are the default values:
ServerLimit 256
MaxClients 256
MaxRequestsPerChild 4000
Thanks,
Each user should only have a single connection to the XMPP server if your using something like BOSH to make the Web->XMPP connection. Each MUC room activity will be sent to the XMPP server with the user's JID as the target and that will cause the server to send that to the appropriate BOSH connection. Same for presence.
The information you need to load balance the Apache setup is the same as load balancing any long-polling connection - XMPP doesn't add anything exotic to the mix. An XMPP BOSH connection is a long-polling connection to your web server which is then sent to a persistent connection to the XMPP server - causing 2 sockets to be opened per user connect.
I'm guessing this a pre-fork Apache....
If you've got 1000 concurrent chat connections, then you need at least 1000 webservers (presumably you'll be serving up static content too, and you say that each presence results in 2 connections - so you can double the 1000) - a ServerLimit/MaxClients of 256 is not going to cut it, you probably need around 2200 to support this (but without hard metrics its hard to give an exact figure).
This is a ridiculously large amount. To support this I'd be looking for 3 boxes, each with about 2Gb free memory before the webserver is started.
MaxRequestsPerChild is not really relevant other than the fact you want some turnover of processes, particularly if you are using long polling.
This is one of the reasons why COMET is not a good idea. Using AJAX polling would be much more efficient. Assuming this is not something you can change, you might want to have a look at using a threaded Apache webserver which is marginally more memory efficient,
C.