vicidial.org

I am working with a company using vicidial using Zoiper and/or x-lite client. I am running a Mac with excellent broadband connection.

The problem I have is that every call I make through vicidial is virtually inaudible due to the amount of crackle. I can hear who I am connected to but they can only just about here me.

I have tested a direct dial 'pipe' set up by the service provider and had excellent sound quality. I use Skype and other VOIP systems without any sound quality issues.

It just seems to be when I am trying to connect/dial through vicidial.

Support from the provider has dried up.

Anyone have some suggestions to try and resolve this?

Thanks in advance

Have you done a ping test to the carrier's IP and checked for lost packets?

Have you done a jitter test and general voip test? http://myspeed.visualware.com/indexvoip.php

Let us know your results.

Have carried out ping test which was good.

0.0% Packet loss

Jitter test is showing...

Test Type: Voice over IP
Downstream Jitter: -1ms
Downstream Loss: 0 %
Downstream Packet Order: 0.0 %
Upstream Jitter: -1ms
Upstream Loss: 0 %
Upstream Packet Order: 0.0 %
Packet Discards: 0 %
MOS Score: -1.0

There is however a comment...

MVLS03: VoIP packets are arriving out of order.

Delivering quality voice over the network requires all packets sent to arrive in a timely fashion. If packets are delivered out of order for any reason then this can cause the speech quality to decline. For example if a packet for the beginning of a sentence arrives in the middle sentence then it is likely that the packet will be too late for acceptable usage and will be discarded. The extent of the quality loss will depend on how many packets are out of order, early or late, and as a results become unusable in the context of uniform speech.

Now, there is a problem with using the link you gave. I am in Portugal and there is no available test centre in Portugal OR Spain for that matter so the test was carried out via Paris, France. I know there are servers in bot Portugal and Spain but just not available on the link you gave.

Also, I have no problem whatsoever using Skype or other VOIP products on the same network.

Hope that helps a little bit?

I believe those proprietary VoIP softwares like Skype will use their own audio codecs that have various measures in place to make call quality "better" than SIP and IAX2. After all, they are designed to be used on a mass scale by people using a large variety of internet infrastructure and little to no technical knowledge... the tradeoff likely being an artificial increase in audio lag and a higher bandwidth requirement.

IAX2 and SIP are protocols that are a lot "closer to the metal" and thus will be more susceptible to any issues between you and your endpoint - IAX2 in particular, as I believe SIP protocol waits for the entire audio "packet" to arrive before assembling it in order and transmitting it (any telecom experts please expand on this or correct me if I'm wrong - I left my "VoIP Deployments for Dummies" book at my last workplace!).

I would say the packets arriving out of order could certainly be a factor for you here. Just at a first diagnosis, I would check my intra-office infrastructure if I were you: sounds almost like a bad router or switch that's causing you heartache. I had problems as well, using an old consumer Linksys router that was on its last legs. It just couldn't cope with the volume of traffic I was putting through it inside the LAN as well as the traffic going out from ViciDial afterwards.

Have you tried having another VoIP provider?
Have you tried using another headset?
Are you behind NAT?
Did you try calling extension to extension locally and still get a crackling sound?
If your using x-lite or zoiper did you disable the acoustic echo cancellation or any other software based echo cancellation?
Try doing a mtr test from your server to your voice provider

1) Welcome to the Party! (Even if I'm late, you're still welcome. LOL)

2) when you post, please post your entire configuration including (but not limited to) your installation method and vicidial version with build.

this IS a requirement for posting along with reading the stickies (at the top of each forum) and the manager's manual (available on EFLO.net, both free and paid versions)

You should also post: Asterisk version, telephony hardware (model number is helpful here), cluster information if you have one, and whether any other software is installed in the box. If your installation method is "from scratch" you must post your operating system and should also post the .iso version from which you installed your original operating system. If your installation is "Hosted" list the site name of the host.

If this is a "Cloud" or "Virtual" server, please note the technology involved along with the version of that techology (ie: VMware Server Version 2.0.2). If it is not, merely stating the Motherboard model # and CPU would be helpful.

Similar to This:

Vicibox X.X from .iso | Vicidial X.X.X-XXX Build XXXXXX-XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation | Intel DG35EC | Core2Quad Q6600

3) Baby steps ... stage one: Is the problem with Asterisk or Vicidial or ...? Let's try our first simple test: Dial through the Vicidial server without using Vicidial, and then again using Vicidial. One dial 91XXXXXXXXXX directly from the soft phone (not in vicidial at all) or the proper dial pattern based on your carrier setup (preferably your cell phone so you can easily hear the quality) and then log in as an agent and use "manual dial" within the logged in agent screen to dial the same number (without the 91 ... because the Dial Prefix for the campaign should provide this automatically). See if one or both of those calls have clear quality.

Also check that your traffic for SIP and RTP are in UDP (not TCP).