How many parallel lines does a provider's SIP account have?

Question

Asterisk

Hello. The situation is as follows. There is a call center that makes outgoing calls in several directions. Let this be Ufa, Moscow, St. Petersburg, Yekaterinburg, Chelyabinsk. from 100 to 400 simultaneous connections and calls.
We want to install a server on Asterisk and minimize communication costs in different directions.
I looked in the direction of MangoTelecom, very high rates in Russia, but they provide 100 lines for each account.
And how many lines are provided by other providers?
For example, we will connect different providers in different directions.
Let's say in the city of Ufa it is cheaper to use a trunk of a local provider with a subscription fee for physical lines
But what if you use the same SINET for outgoing calls to other cities? How many parallel outgoing calls can be made from one account? How to properly implement call distribution? Are there any restrictions for providers? After all, in fact, this is telephony over the Internet; there should not be a limit on the number of lines.

Answer 1

Don't listen to the theorists. 400 simultaneous lines on the asterisk is not a problem, but only if there is no transcoding, and there is no processing of the voice stream on the side of the asterisk itself.
Those. if the options for managing redirection are removed from the Dial, no inband dtmf (but, fortunately, no one in their right mind usually uses it anyway), and an identical list of codecs for subscriber devices and uplink, it will work without problems.
But with network cards there can be a rake, it is necessary to test, and in general, the performance solution should, of course, be tested before commissioning.
www.voip-info.org/wiki/view/Asterisk+dimensioning - read this. Thousands of simultaneous connections for an asterisk on modern server hardware are no longer a problem.
And it is worth remembering that between the major versions, the asterisk had significant performance improvements.

Answer 2

By the way, a system that will consistently serve 400 simultaneous calls is a very cool system, this is an operator level. You are unlikely to build it yourself, unless you have vast experience in building such systems.
For example, a ready-made boxed solution based on Asterisk from Digium (Asterisk developers), Switchvox - 75 simultaneous calls are promised for the older model. You will need three of these pieces of iron to serve your expected load level.

Answer 3

In Sipnet - as much as you like. At least we didn't get to the limit when we tested it.
Each conversation will be billed independently.
In general, it depends on the provider and is prescribed in the contract.

Answer 4

100% CPU utilization on the server is generally evil. Especially for realtime tasks (which include IP-telephony).
With faxes - I agree, everything depends on the tasks. When I made large systems for offices, faxes were served separately (up to connecting to separate gateways that directly contacted the operator).
But I successfully made more than 900 simultaneous connections in 1 server. There were no complaints about the quality of communication.
Again, I agree with the fact that everything depends on the specific tasks and required services.
Fortunately, telephony is perfectly clustered.
Specifically, in the original task, the solution is obvious - to clamp G.729 as the required codec for all peers (because all trunks usually know how to do it, but not everyone knows how to G.711) - thereby get rid of transcoding. I think even transfer will work under such a load, although it needs to be checked.
If you need to record negotiations (and it will probably be needed in the call center), then there will be an ambush - because 400 write streams to the disk will completely kill the disk subsystem. So you will have to write to tmpfs, and then transfer it to disks in one stream. In addition, you will have to write directly raw G.729, and how and where to recode it into a format suitable for listening is another story.
In general, there is nothing unsolvable here. This is not an easy task, requiring the selection of equipment, tests, etc. - but completely manageable.

Answer 5

Try MultiFon.
Megafon has now begun to provide SIP trunking services. The number of parallel incoming/outgoing is configurable.

Answer 6

400 simultaneous conversations for asterisk is not a problem if:
1) Packet2packet bridging (sip to sip, h323 to h323) is used, for example, when RTP packets are processed in the RTP stack and asterisk does not allocate a separate thread for processing it;
2) MOH is not used - in this case asterisk acts as a voice stream mixer, and it uses transcoding to its internal slin format (you can see asterisk -rx 'core show translation';
3) In-band DTMF is not used.

Answer 7

Comrade mithraen, cisco 3845, and more specifically CUCME, this is a gateway of pure water, it does not “pull” the role of a sip proxy, you can use similar functionality in CUCME, it is called CUBE.

Answer 8

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