PBX VOIP Call quality problem & How to troubleshoot?

[oceans11]

Hi there,

We have a new PBX VOIP phone system which is causing me a lot of stress.

The set up is as follows:

Telkom line 4mbps

Planet 24-Port 10/100Mbps Fast Ethernet Switch

PBX Server running VMware
Elastix (latest version) / Asterisk / FreePBX
IAX trunk to Switch Telcom http://www.switchtel.co.za/
Intel Core i3-2100 CPU @3.10Ghz
Dedicated 2 gig ram
ZyXEL Fast Ethernet PCI Adapter FN13 100/200Mbps

We are using 4 Counterpath X-light softphones

There is more than 60% of the server and PBX resources free.

Our call quality is not good. The best example I can give you is if we are giving a phone number to someone on a call, they may miss a number because the call skips or breaks for half a second.

I am at a loss as to where to start to fix this problem. Any advice would be appreciated.

One thing to note is we need to have our pbx hosted locally in our office. If I hear my employees say "can you hear me" to our client one more time, I may jump through the window of my office!

 

[InfinityMVS]

What codec are you using ?

Why have you opted for the softphones?

Call me on 021 200 1780 I'm running VoIP with right codec on Trixbox and will be more than willing to help you.

Cheers

 

[oceans11]

What codec are you using ?

Why have you opted for the softphones?

Call me on 021 200 1780 I'm running VoIP with right codec on Trixbox and will be more than willing to help you.

Cheers

Thank so much for your time and info. I will let you know how I get on.

 

[mo_to]

Hi

Your ADSL connection, who is your ISP, and whatping times do you get connecting to the switchtel server?

Also, how congested is your ADSL network?

 

[gmza]

You didn't mention the "PBX Server running VMware" when speaking to me (Switch Telecom).

Asterisk is an extremely time-sensitive application. Virtualising it can be done, but is not recommended unless you can be sure that you're allocating sufficient CPU processing power and that your timing is not going to be affected.

Asterisk IAX2 trunking takes timing from the DAHDI driver. run dahdi_test to ensure that you're getting reliable timing, particularly in light of the virtualisation.

If need be, convert to SIP and/or move from a VM to a dedicated machine.

 

[oceans11]

Hi

Your ADSL connection, who is your ISP, and whatping times do you get connecting to the switchtel server?

Also, how congested is your ADSL network?

Hi. My adsl connection is a 4 meg line, getting about 0.45 up. Data provided by Afrihost (capped package) I am not sure how to find out how congested the local ADSL network is.

 

[oceans11]

You didn't mention the "PBX Server running VMware" when speaking to me (Switch Telecom).

Asterisk is an extremely time-sensitive application. Virtualising it can be done, but is not recommended unless you can be sure that you're allocating sufficient CPU processing power and that your timing is not going to be affected.

Asterisk IAX2 trunking takes timing from the DAHDI driver. run dahdi_test to ensure that you're getting reliable timing, particularly in light of the virtualisation.

If need be, convert to SIP and/or move from a VM to a dedicated machine.

Hi correct, I am sorry I did not mention that. I am new to this.

This is what I get from the test

dadhi_test output is:
Opened pseudo dahdi interface, measuring accuracy...
99.772% 99.772% 99.772% 99.250% 99.772% 99.772% 99.772% 99.772%
99.250% 99.772% 99.772% 99.772% 99.250% 99.772% 99.772% 99.772%
99.250% 99.772% 99.772% 99.772% 99.772% 99.250% 99.772% 99.772%
99.772% 99.250% 99.772%
--- Results after 27 passes ---
Best: 99.772 -- Worst: 99.250 -- Average: 99.655898, Difference: 99.989231


[root@blackthorn dahdi]# dahdi_test
Opened pseudo dahdi interface, measuring accuracy...
99.862% 99.862% 99.862% 99.862% 99.861% 99.862% 99.159% 99.862%
99.862% 99.862% 99.861% 99.862% 99.862% 99.159% 99.862% 99.861%
99.862% 99.862% 99.862% 99.862% 99.159% 99.862% 99.862% 99.862%
99.862% 99.862% 99.862% 99.159% 99.862% 99.862% 99.862% 99.862%
99.862% 99.862% 99.163% 99.822% 99.822% 99.822% 99.822% 99.822%
99.200% 99.822% 99.822% 99.822% 99.822% 99.200% 99.822% 99.822%
99.822% 99.822% 99.822% 99.200% 99.822% 99.822%
--- Results after 54 passes ---
Best: 99.862 -- Worst: 99.159 -- Average: 99.747977, Difference: 99.992457
[root@blackthorn dahdi]#


PING TEST

[root@blackthorn dahdi]# ping sip.switchtel.co.za
PING rba2.switchtel.co.za (196.38.164.132) 56(84) bytes of data.
64 bytes from rba2.switchtel.co.za (196.38.164.132): icmp_seq=1 ttl=56 time=70.1 ms
64 bytes from rba2.switchtel.co.za (196.38.164.132): icmp_seq=2 ttl=56 time=60.1 ms
64 bytes from rba2.switchtel.co.za (196.38.164.132): icmp_seq=3 ttl=56 time=60.1 ms
64 bytes from rba2.switchtel.co.za (196.38.164.132): icmp_seq=4 ttl=56 time=90.1 ms
64 bytes from rba2.switchtel.co.za (196.38.164.132): icmp_seq=5 ttl=56 time=80.1 ms
64 bytes from rba2.switchtel.co.za (196.38.164.132): icmp_seq=6 ttl=56 time=70.1 ms
64 bytes from rba2.switchtel.co.za (196.38.164.132): icmp_seq=7 ttl=56 time=80.1 ms

--- rba2.switchtel.co.za ping statistics ---
7 packets transmitted, 7 received, 0% packet loss, time 6011ms
rtt min/avg/max/mdev = 60.116/72.998/90.174/10.321 ms

 

[oceans11]

Now im getting this:

[root@blackthorn ~]# dahdi_test
Opened pseudo dahdi interface, measuring accuracy...
99.949% 99.949% 99.949% 99.949% 99.949% 99.949% 99.949% 99.949%
99.949% 99.949% 99.949% 99.949% 99.949% 99.949% 99.949% 99.949%
99.949% 99.070% 99.949% 99.949%

 

[paulcolmer]

I have never been a great fan of running Asterisk in VMware, the only thing that Asterisk likes sharing with is the open source community not the processor that supports it. And i concur with GMZA with regard to changing to SIP G729, this can be changed by Switchtel, you could even try a free G729 codec for testing but have found at times that the Digium codec performs better. There is no real need for using IAX2 with only 4 phones, your 450kb/s upload can handle G729 unless you got big data on the same DSL, look at the QoS in the router. If all else fails go to a dedicated server, its the best option for any flavor of Asterisk.

 

[oceans11]

I have never been a great fan of running Asterisk in VMware, the only thing that Asterisk likes sharing with is the open source community not the processor that supports it. And i concur with GMZA with regard to changing to SIP G729, this can be changed by Switchtel, you could even try a free G729 codec for testing but have found at times that the Digium codec performs better. There is no real need for using IAX2 with only 4 phones, your 450kb/s upload can handle G729 unless you got big data on the same DSL, look at the QoS in the router. If all else fails go to a dedicated server, its the best option for any flavor of Asterisk.

Thank you for your comment. We may well go for the dedicated server as we are growing quickly and will need more lines.

 

[werner]

to add my 2c

we ran asterisk in a vmware esxi very powerful server, had call breakups...timing issue, elastix has a specific kernel you can try...helped a bit but not resolved.

moving to g729 also helps, less bandwidth etc...QOS helps...same reason.

what will still stuff you over is xlite, which is running on a pc, which has an ethernet connection...so while your agent is using the pc to update call records/raise a support ticket etc, this moving data/bandwidth is interfering with the critical time sensitive voip packets.

same if you get physical phones, a lot of them allow you to run it in bridge mode so that you can daisy chain your pc on the end of the phone....most likely a certifiable bad move with regards to call quality...

keep your voice and data network seperate as much as possible...vlans, physically different switches etc. dedicated adsl...use a physical server so that you have a reliable timing source...then you will be cooking.

 

[mo_to]

Hi

How many concurrencies do you have?

These are the hurdles you currently face:

1- Asterisk on VMware, not good. You need a dedicated server. If you want a much easier and professional approach, visit www.1cloud.co.za . its a Cloud Hosted PBX which will save you alot of headache and money. There is a 15day trial.

2- You ADS Upload is .45, its actually 225kbps...always half it. with congestion, you will have some quality issues, but good enough to run a small business.

3- Your ping times of 60ms is not good. If you're in JHB, you should be getting 10ms, and if another province, in the region of 30ms to 40ms. Latency is EXTREMELY important for good quality VoIP.

4- You are running the voice over the internet cloud. The packets reach the internet traffic, loses quality and potential break up..

All in all the whole design needs an overhaul. I would try 1cloud.co.za which will def fix your issues, as the voice packets reach the switches via a private national ADSL MPLS. It's at least 6x higher quality and more stable.

 

[oceans11]

to add my 2c

we ran asterisk in a vmware esxi very powerful server, had call breakups...timing issue, elastix has a specific kernel you can try...helped a bit but not resolved.

moving to g729 also helps, less bandwidth etc...QOS helps...same reason.

what will still stuff you over is xlite, which is running on a pc, which has an ethernet connection...so while your agent is using the pc to update call records/raise a support ticket etc, this moving data/bandwidth is interfering with the critical time sensitive voip packets.

same if you get physical phones, a lot of them allow you to run it in bridge mode so that you can daisy chain your pc on the end of the phone....most likely a certifiable bad move with regards to call quality...

keep your voice and data network seperate as much as possible...vlans, physically different switches etc. dedicated adsl...use a physical server so that you have a reliable timing source...then you will be cooking.

Thanks for the comprehensive info. Yesterday we had a seperate ADSL line for the PBX installed and have seperated the data and voice network. We have also moved from IAX to SIP. Call quality is much better now. I think in time we will be using a dedicated server for the PBX.

 

[oceans11]

Hi

How many concurrencies do you have?

These are the hurdles you currently face:

1- Asterisk on VMware, not good. You need a dedicated server. If you want a much easier and professional approach, visit www.1cloud.co.za . its a Cloud Hosted PBX which will save you alot of headache and money. There is a 15day trial.

2- You ADS Upload is .45, its actually 225kbps...always half it. with congestion, you will have some quality issues, but good enough to run a small business.

3- Your ping times of 60ms is not good. If you're in JHB, you should be getting 10ms, and if another province, in the region of 30ms to 40ms. Latency is EXTREMELY important for good quality VoIP.

4- You are running the voice over the internet cloud. The packets reach the internet traffic, loses quality and potential break up..

All in all the whole design needs an overhaul. I would try 1cloud.co.za which will def fix your issues, as the voice packets reach the switches via a private national ADSL MPLS. It's at least 6x higher quality and more stable.

Thank you. 1cloud.co.za is unfortunately not going to work for us, because we need the pbx to be hosted locally.
We are based in Cape Town, but I have a feeling our exchange is somewhat overloaded.

 

[kilos]

We are based in Cape Town, but I have a feeling our exchange is somewhat overloaded.

Purchase an Unshaped ADSL account
Check your have low latency (<10ms)to the 1st DSL hop after your DSL router,

Try the free Telkom ADSL Guest account "guest@telkomadsl" Password: guest
Ping www.telkom.co.za, if Guest account is incorrect call Call center 0800375375

Interesting discussion on Digium forum running Asterisk on Vmware timing issues that gmza (Greg) mentioned earlier on.

 

[mo_to]

Thank you. 1cloud.co.za is unfortunately not going to work for us, because we need the pbx to be hosted locally.
We are based in Cape Town, but I have a feeling our exchange is somewhat overloaded.

Contrary to your belief, 1cloud.co.za is hosted locally, with presence at JINX, CINX, TERACO and others.

The ADSL is also on a private MPLS, so you do not need bandwidth, nor will it cost you any bandwidth. it's like connecting your hosted PBX on a massive LAN. Latency to 1cloud is approximately between 8ms and 40ms running off the private MPLS. You wont go wrong.. 100% sure of it!

 

[werner]

if you ivr's still sound choppy/compressed/spoken too fast then it is a vmware issue, and you would do well to install a kernel specifically for vm machines...otherwise, enjoy it!

 

[Superspeed]

We had the same problems. Solved it by getting USB headsets instead of analogue ones. By using a softphone you are going to get issues if you don't use USB headsets. The main reason is that ANYTHING that runs on your PC will affect the voice e.g. anti-virus, defragmenter etc. Give it a go