OS and BSS Optimization Recommendations

[skb007]

1. Brekeke Product Name and Version: BSS Adv Ed 3

2. Java version:1.8

3. OS type and the version:RHEL 7

4. UA (phone), gateway or other hardware/software involved:Various

5. Your problem: Need recommendation for OS as well as BSS optimization.

Server Specs: Dual 6 core processors.
48GB RAM

We are expecting 100CPS with no media and 5000 simultaneous calls.

OS: Any recommendations for optimizations at OS level. I have already increased the file descriptors.

TOMCAT: Anything needs to be optimized at Tomcat level. We have installed Tomcat 8.5.34 with default configuration except autoDeploy="false" and liveDeploy="false".

BSS:
Config->System->Java: Anything should go in the Java VM aruments ??
Config ->SIP-> Performance Optimization Proxy : Initial threads 10 and max sessions per thread 50.

Any recommendation would be much appreciated.

[Haddas]

> OS: Any recommendations for optimizations at OS level.

Is it running on a physical machine? or virtual machine?


> TOMCAT: Anything needs to be optimized at Tomcat level.

Tomcat is used as web server. It doesn't affect call performance.


> > Config->System->Java: Anything should go in the Java VM aruments ??

At this time, you don't have to.


> Any recommendation would be much appreciated.

Does your customer send SIP-REGISTER?
Are you using SIP-Authentication for INVITE and/or REGISTER?
How many DialPlan rules do you have?

[skb007]

System is running on physical server.
No SIP register. We have dial plan configured to send 606 for REGISTER.
NO SIP Authentication for INVITE.
Currently, we have 10 dial plans but it may increase to 20-30 eventually.


I am also doing some logging in dial plan such as following. Is this going to cause some issues?

Matching Patterns

$request = ^INVITE $rule.sps = (.+)

Deploy Patterns

log = STATS:%{$date} %{$time},"BREKEKE",%1,%{$sessionNum}

[Haddas]

Does the SIP Server connect to a SIP provider?
Does the SIP Server connect to a VoIP gateway?
Who is handling RTP packets?

> I am also doing some logging in dial plan such as following. Is this going to cause some issues?

Is it for statistics purpose?
Since it adds new log line to the log, a little (ignorable) overhead might be introduced.

[skb007]

Here is the call flow.

Customer --->BSS-->Provider

There is no Authentication on either side. Both sides are using firewalls so only trusted IP's can reach BSS.
There is no VOIP gateway involved.
RTP does not flow through the BSS. It will go directly between customer and provider.

[Haddas]

It seems there is no problem because your deployment is simple.