2012 VoIP With Virtualization

Hi FS Users, 

What is the consensus about using virtualized servers for real-time voice (RTP)?  Even up until a few years ago it was hard to guarantee the CPU cycles to the voice nodes. 
Virtualizing the signalling (SIP) has always been favourable in terms of HA. 

If the entire host is under control, is it safe to say - we can virtualise voice? 
Does it depend on what else is shared in the host - say it would be silly to put a high load database server on the same hardware node... 

Thanks for your inputs 

Chris 

--------------------

basically it depends ... on MANY things..

a) how much you care about your voice quality

b) how many concurrent calls 

c) do you modify the RTP stream or just pass it through.

A）你有多么在乎你的声音质量

b）有多少个并发呼叫 

c）你修改的RTP流，或者只是把它传递

basically if your real picky or have more than a handful of concurrent calls.. then dont do it.

if your using conferences etc... dont do it.

基本上，如果你真正的挑剔或以上的并发调用了一把.. 不这样做。

如果您使用的会议等。.. 不这样做。

that being said, you can get away with a lot if you dont mind the occasional blip and chirp in your media..

这样说，你可以走了很多，如果你不介意偶尔在你的媒体的消息，并啁啾..

Thing to not though... above is for virtualization ... ( ESX, Xen, etc )

Jails like OpenVZ, BSD Jails, etc work well and provide many of the benefits found in virtualization.

Jay

-------------------

Hi, Chris

That depend of your enviroment. First you have to take a sort of scientific approach: define, trial and test again and again until the result be consistent.
Second tell us about your enviroment:
    - Real hardware: the server that will run your vm. ¿Are blades, small server, etc? VM software used (VMware, OpenVZ, XEN, Jails...)?
    - ¿How you set your VM? cpu, cpu cores, memory, disk, #nic?
    - ¿How many calls have you handle in the real world?
    - ¿Transcoding?
    - ¿Have you tested your VM?

-硬件：服务器将运行你的虚拟机。¿刀片，小型服务器等？VM使用的软件（VMware的，OpenVZ的，XEN，监狱...）     

- ¿如何设置你的虚拟机呢？CPU，CPU核心，内存，磁盘，＃网卡？     

- ¿在现实世界中，你有多少呼叫处理？     

- ¿转码     

- ¿你测试你的VM虚拟化环境的VoIP的增长。

Virtualized enviroment for VoIP are growing. I am consultant for contact center services, and I have receive a lot of questions about virtualization. A solution provider can give some guidence about sizing and technical considerations but at the end the final word is:  YOU HAVE MAKE A LOAD TEST TO TEST YOUR ENVIROMENT AND IDENTIFY POSSIBLE ISSUES BEFORE GO TO PRODUCTION!!!!

Virtualization solutions will  do their homework to make VoIP a secure option. Look this paper for wmware:
http://blogs.vmware.com/performance/2012/01/voip-performance-on-vsphere-5.html
http://www.vmware.com/files/pdf/techpaper/voip-perf-vsphere5.pdf

--------------------------------------

tl;dr;  3 opportunities, none are urgent.  virtualization isn't a need-to-have but it would be nice to know what is do-able these days... i've had the best part of 4 years away from 100% voip, and spent 2 years in academia on gentoo plus haven't worked for nearly 12 months just faffing about with J2EE stuff and a bit of recent asterisk/kamailio and now freeswitch.

Hi Dario - I am creating a load test at the moment for a queue of 90 calls and comparing FS with asterisk for queuing - just reading over my old notes for SIPp.  I am very familiar with asterisk through to 1.4 but have been doing plain old web sys admin with gentoo in academia for 2 years and stayed away from voip and the industry - but times/opportunities are changing up to get back into it.   I have lurked in the FS irc channel for about 6 months whilst doing this other stuff and now is the time to pull the finger out so to speak.

Current opportunities are 2 potential network providers and a start up i'm the tech guy for:

A) This 90 call queue scenario will not be virtualised as it probably would not suit queuing which I understand to be a bit hungry for resources - it will be a failover set up for an Avaya where the avaya is on-site and this hosted-queue will be at the DC.

B) I am working on another startup that does purely conferencing and sms (hence my persistence with mod_gsmopen over the last couple of days)

C) Am getting some information on a hosted pabx environment where the current set up is 2 asterisk boxes with 700 registrations and about 60 concurrent calls and where to go with it.  the people that run that show haven't told me too much  - i.e. we want to give access to "IT Guys" to administer their own customers, we just want to bill the minutes  VS  they have their own hosted apps and want to keep everything 100% theirs.

A）这90个呼叫队列的情况下将不被虚拟化，因为它可能不适合排队，我理解是有点饿了的资源 - 这将是成立的Avaya Avaya是现场和本托管故障转移队列将在DC。

B）我的工作再次启动，纯粹会议和短信（因此我的持久性与mod_gsmopen在过去的几天）

C）我得到的一些信息在托管程控交换机环境下，目前的设置是2个星号箱700注册和大约60个并发呼叫和去用它。运行该节目的人，都没有告诉我太多 - 也就是我们想给获得“IT人”来管理自己的客户，我们只是想账单分钟与他们有自己的托管应用程序，并希望把一切都他们的100％。

The great thing with (C) is that I will be doing system engineering and no support.

I will read through the links you sent.  I am unfamiliar with virtualisation in terms of real world usage / performance etc. but have dabbled in a convenience-for-development  xen and virtualbox.

Using openvz in a "you can manage your own customer handsets" is appealing - i just had a quick read of it all.

But yes -- it's a load testing thing at the end of the day, and there can be no blips, or other voiping artifacts (i.e. digitized voice and the like)

In general:

1.  No transcoding (ever) - it will all be alaw all the way.

2.  All servers have Xeon CPUs.  Whether they are recent or from the last 5 years is another story depending on the availability of servers and/or cash.  I think there are more servers than cash to spend - rackspace is not a problem.  That being said, I just priced up some dells and supermicros to have some kind of figures ready.

3.  Currently, there are 60 concurrent calls across about 700 registrations.

1。无转码（永远） - 它都将是alaw的方式。

2。所有服务器都Xeon处理器。无论他们是最近从过去5年则是另一回事，这取决于服务器的可用性和/或现金。我想有更多的服务器比现金支出 - Rackspace公司是没有问题的。话虽这么说，我只是售价高达一些峡谷和supermicros的有某种数据准备的。

3。目前，有60个并发通话的约700注册。

Switches and routers are none of my concern, but they use a range of some cisco switches, a redback router, extremenetworks switch and some juniper firewalls will be in before end of year.

There is no rush on any of the above, it's all early days, plus I still have some of this non-computer project work I've been doing - manual labour upgrades...fun fun fun!!!

Thanks for your links and comments - I look forward to hearing more.

----------------------------------------------

I currently run a testing setup on rackspace cloud, doesn't seem to be too bad, but then again the call volume is low.

----------------------------------------------

For the setup in question:

 

The hardware:

IBM SystemX 3650.

8 core Xeons, E5*.

48GB RAM

Virtualization is done by VMWare ESXi, mostly 4 or 5.

Network is all Cisco.

 

3 VM Boxes that reach an average of 400 calls during working hours.

FreesSWITCH there is very old(1.0.6).

4 CPU Cores

2GB

64 bit Centos 5.2

local dialplan XML

no transcoding

xml_cdr

very stable.

 

Servers have several virtual machines with a lot of different things running, FSs have priority on hardware resources...

 

We have some other VM boxes(FS Sep 2011) with less load(270 calls avg) that do trancoding(25 calls avg g729<->g711), no quality issues but the FS clock drifts a lot(up to 10 minutes a day), VMware's fault ->  http://www.vmware.com/files/pdf/Timekeeping-In-VirtualMachines.pdf

fsctl sync_clock solves the problem but might ruin CDRs so BEWARE!

 

There are several other VMs in quite a few hardware setups, but none of them has impressive call volumes...

 

P.S.: Please don't shout at the age of the installs... they have served us well! :D.

硬件：

IBM SystemX 3650。

8核心Xeon处理器，E5 *。

48GB的RAM

虚拟化是VMware ESXi的，多为4或5。

网络是所有的思科。

 

3 VM盒，在工作时间内达到平均400电话。

FreesSWITCH是很老的（1.0.6）。

4个CPU核心

2GB

64位元的CentOS 5.2

本地拨号方案的XML

不转码

xml_cdr

非常稳定的。

 

服务器有多个虚拟机运行了很多不同的东西，FSS对硬件资源的优先级...

 

与负载（270呼叫平均），trancoding（25个电话平均G729 < - > G711），无质量问题，但FS时钟漂移了很多（最多10分钟的时间，我们有一些其他的虚拟机盒（FS 2011年9月） ），VMware的故障- >  http://www.vmware.com/files/pdf/Timekeeping-In-VirtualMachines.pdf

FSCTL sync_clock解决了这个问题，但可能会毁了的CDR所以要小心！

 

还有几个其他的虚拟机在相当多的硬件设置，但他们没有令人印象深刻的通话量...

As said before, it depends heavily in your setup... We do virtualize FreeSWITCH and aside from a few glitches it has been good... We have approx 400 calls per VM. Later when I hit the office I can describe my setup in detail.

正如前面所说的，它在很大程度上取决于你的设置...  我们做虚拟化FreeSWITCH的，除了一些小问题，它已经好了... 我们每一个虚拟机有大约400电话。 后来，当我打的办公室，我可以描述我的详细设置。

http://freeswitch-users.2379917.n2.nabble.com/2012-VoIP-With-Virtualization-td7559541i20.html