1 Agent Server+ Balance Dialing VS Agents On Many Dialers

[vkad]

We are trying to load balance the Dialer. We are using WebRTC + Viciphone

This server has 12-15 agents and peak calling is at 10 calls per agent being dialed.
At a peak of 170 - 180 calls being dialed the audio becomes CHOPPY even though the server load is only at 50%.

We started out as vicibox-express on this install.
We need to expand the cluster.


We are faced with two choices, either to have:
==================================================================
Asterisk Dialer act as agent server, with 120 channels allowed for dialing (AGENTS ALLOWED ONLY ON THIS SERVER) (Works as the WebRTC server)
+ An Additional Dialer with another 120 channels allowed for dialing through BALANCE DIALING (NO AGENTS ALLOWED ON THIS SERVER) (No need to enable WebRTC on additional Dialer as calls are given to the agent asterisk server)
=========================================================OR================================================================
Asterisk Dialer act as agent server, with 120 channels allowed for dialing (HALF AGENTS ON THIS SERVER)
+ An Additional Dialer Asterisk Dialer act as agent server, with 120 channels allowed for dialing (HALF AGENTS ON THIS SERVER) (How will this server work with WebRTC?)
==================================================================

Vicibox 8.0.1 with Asterisk 13.20.0
Remote Archive Server
1 x Express = E3-1270v5 + 16GB DDR4 + 256GB SSD (Current)
+
1 x Dialer = E3-1270 + 8GB DDR4 + 128GB SSD (Additional Server for Dialer we would like to add)

[mflorell]

We usually recommend spreading your agents out and using Phone Aliases so you can use phone login load balancing to both balance your agents across your telco servers, as well as offer a failover option if one of the telco servers goes down for any reason.

[vkad]

We usually recommend spreading your agents out and using Phone Aliases so you can use phone login load balancing to both balance your agents across your telco servers, as well as offer a failover option if one of the telco servers goes down for any reason.

Thanks for the advise. If I use a single web server with Phone Alias, will the agents be automatically load balanced to different dialer servers?

Additionally, will phone aliases work with WebRTC (Viciphone), since we need to specify different tls certificates for each dialer server.


IF we go ahead with single agent server and multiple Dialers for load balancing, will there be a channel limit on the agent server?

[mflorell]

Yes, Yes and there is always a channel limit for servers that outbound dial, you just need to figure out what an acceptable limit is for your specific setup.

[vkad]

Yes, Yes and there is always a channel limit for servers that outbound dial, you just need to figure out what an acceptable limit is for your specific setup.

Thank you for the quick reply.

I understand that each dialer has a limit of channels, now if I was to use a single agent server (with lets say 120 chaneels and 20 agents) dialing 1:10 and I add another server for balance dialing and setup 150 channels on this for dialing. Will the first server take up channels to fill/balance calls with the other server.
If the calls are being dialed on the outbound dial on the second server while balancing these dials for the first server, will the channels on the first server be used up as well.

In other terms, if server 1 was balacing 120 calls to server 2, will server 1's 120 channels be used as well or will it only be for server 2?

[mflorell]

Calls are placed based upon where the agents are logged in first, so in a load-balanced setup, it should be fairly even across all of your dialers.

[vkad]

Calls are placed based upon where the agents are logged in first, so in a load-balanced setup, it should be fairly even across all of your dialers.

Do balanced calls consume channels on both servers or just 1 server?

For example 1:10 ratio calls from server 1 balanced on server 2 will use 10 channels on server 2 only or use 10 channels each on both of the servers?

[williamconley]

Channel usage:

1) Agent logged in to the meetme room. Every agent consumes one channel while logged in at all times. No exceptions.

1 Channel per agent logged in

2) Outbound call to Prospect. Whether or not they "answer" a channel is used from the moment of dial to the moment of hang up for each dialed number.

1 Channel per dialed prospect

3) Cross-Server connections when the Prospet call and the Agent call are on different servers: This occurs when a Prospect answers, and the agent to which this call is routed is on a server other than the one that dialed the prospect. This connection requires creating an extra IAX channel on BOTH servers. This happens more often when "balance dialing" is active because dialers will call when other dialers are low on available channels for a campaign, making it just "more likely" that when a Prospect says "hello", the agent to which they will be assigned will be in a meetme room on another server.

2 Channels per prospect ONLY if cross server is necessary.

[vkad]

Does asterisk generally scale poorly with cross server IAX channels? Is the limit still around 250 channels for IAX channels as well?

[williamconley]

Does asterisk generally scale poorly with cross server IAX channels? Is the limit still around 250 channels for IAX channels as well?

No.

There is no 250 channel limit. You made that up.

[vkad]

Does asterisk generally scale poorly with cross server IAX channels? Is the limit still around 250 channels for IAX channels as well?

No.

There is no 250 channel limit. You made that up.

http://download.vicidial.com/iso/vicibo ... nstall.pdf
" This is because of the manner in which Asterisk scales starts to break down around or above 250 concurrent internal channels."
I am referring to this limit.

[williamconley]

This is because of the manner in which Asterisk scales starts to break down around or above 250 concurrent internal channels

Ahh, I see your issue. That's not so much a limit as a configuration challenge. Some servers have issues, others don't until a higher value. Test it for yourself.

Vicidial/asterisk/mysql/apache have some artificial limitations that must be removed for enterprise level systems. They can't just automatically be removed, because single server smaller systems would implode/freeze and die if they exceeded too many of those barriers on a single server. But one or two roles on the same server when enterprise levels are reached often require removing some of these barriers. Sometimes it's "more nodes/links" of one sort or another. Sometimes it's "clean stuff up faster". Sometimes it's just "gonna need more memory to do this all at once.

But in the end, most find that the pricing breakpoint works better with more smaller cheaper servers rather than huge enterprise servers ... except for the DB (since there can be Only One).

[vkad]

Oh, ok. We have 16gb + E3-1670v6 servers here for telephony. What limits would you suggest we increase to be able to do maybe 500 channels per server. The load is around 15% on 120 channels with amd enabled.

[williamconley]

That depends on the barriers you hit. Raisse your channel limit. Hit a barrier, post the error, and we'll assist you as you raise the bar.

If it were "here's the settings", they'd already be set. It's about how you use the system. We presently have clients who push 400 channels per server. Back when we had custom Gentoo buildouts, which were very streamlined and compiled that way on purpose, we had clients who pushed 600 channels on 4 core systems. Very good bang for the buck there. Before robodialing became illegal. lol

[vkad]

We don't robodial, but what we do have are poor quality leads. A lot of answering machines, NA, etc. Maybe only 10% people actually pick up the phone.

I am getting the error of too many open files in asterisk sometimes, but way before that calls are garbled up. Even though the system shows less than 15% utilisation.

Are there any particular settings I could look out to raise the limit. The DB server is not even pushing 5-10% load at peak dialing.

[williamconley]

the error of too many open files in asterisk sometimes

Please post an actual error. Vague leads to assumptions which leads to wild goose chases.

calls are garbled up

This is a different story entirely. what codec are you using? Is your internet bandwidth limited? How many carriers (that are entirely unrelated to each other!) have you tried during this scenario (ie: when calls are poor quality, switch to another carrier and see if the calls continue with poor quality on the new carrier).

Attack one problem at a time. Run it to ground. Then move on to the next.

[vkad]

the error of too many open files in asterisk sometimes

Please post an actual error. Vague leads to assumptions which leads to wild goose chases.

calls are garbled up

This is a different story entirely. what codec are you using? Is your internet bandwidth limited? How many carriers (that are entirely unrelated to each other!) have you tried during this scenario (ie: when calls are poor quality, switch to another carrier and see if the calls continue with poor quality on the new carrier).




Attack one problem at a time. Run it to ground. Then move on to the next.

I will put the errors here the next time I get them.


1. Using ulaw/alaw
2. bandwidth is 500mbps
3. using plivo and twilio, they ought to be good, but I will test another carrier.

[williamconley]

bandwidth has two directions. you only provided one number. also, with a number that large I have to ask where you got it ... did you run a test or get it from a brochure or ...?

Also while network speed is a factor in call quality, it's not the only one. In voip jitter and latency both play a part and can be introduced by otherwise innocuous networking components such as switches and routers in the path.

Logging bad call quality events (and all data associated, like time/agent/workstation/carrier/server) has been known to (eventually) identify failing or poor quality (or overworked) components.

Happy Hunting

[vkad]

Bandwidth is on a colocated server. I have tested the bandwidth and run ping tests and everything is good. No delays.

I am sure it is asterisk now, since even though not getting any errors on it this time, asterisk just went kaput.
The web interface was showing dialing, but asterisk wasn't responding anymore. No calls on asterisk -rvvvvvv or any error messages.

As soon as the server was restarted, everything was normal again. I have reduced the trunking to 100 each on servers and no issues.
A friend is using the same carrier, plivo, with 500 concurrent calls without issues (but more dialers on weaker servers).

I am just waiting to see an error if this time asterisk dies on me.