Inbound Audio Issues - 3CX SBC

[joshhough]

Hi,

I'm hoping someone can help... We've put in a hosted 3CX system (3cx on Rackspace Windows server - works fine, with SBC on a raspberry PI at the local site).

We have 2 x Cisco SPA504G and then an RTX phone system - one base station, two handsets.

Firstly everything works... calls come in properly and calls can be made. The audio is the problem... When the extensions make/receive calls, audio inbound (to the extension) is intermittent. By that i mean it works sometimes perfectly, but most of the time it cuts out mid sentence and then back in again which isn't ideal.

I have a DrayTek 3200 which i have spent ages on the phone to DrayTek support about setting up QoS but still it hasn't helped. Does anyone have any further thoughts?

Thanks,

Josh

 

[lneblett]

The first thing to look at is your internet connection, you failed to mention its bandwidth. In many cases, such as yours, the call stream may experience jitter which causes the voice to sound clipped or stuttered and if the condition worsens then dropped packets may occur which results in a loss of the voice stream and if too bad...a dropped call.

Once you know the bandwidth, then you should test it to see if you are getting the paid for speed. Try testing to a number of different locations (speedtest.net is OK for the US market) as various test servers will yield different results due to their loading or other conditions; hence why you should test to several in order to get a decent reference. You should test when you know that the office network traffic is minimal.

Next you should ensure that the firmware for the router is up to date and that the PI is at its latest revision. I am not familiar with the particular model of Draytek, but do use others so I assume the feature sets to be similar. The Draytek may have a diagnostic feature by which you can monitor the data usage. This will show each IP that is transiting the WAN and its associated usage of both up and down traffic. It will also show total usage and peak so you can get a relative idea of bandwidth utilization. IT also has a bandwidth limitation feature by which you can effectively throttle or reserve bandwidth for an IP or group. Most devices will take as much bandwidth as they can or what the opposing device can deliver. You may want to put a reservation in for the 3CX system so as to ensure that enough is always available, Then, in their latest firmware release in January, they have a feature that can be checked for VoIP. This feature reportedly detects SIP calls based on the port usage (5060 or as specified) and then examines the packets to determine the RTP ports/stream and then applies a QoS automatically. I have enabled the feature on a couple of devices and while it does require AIP ALG to be enabled, it does not seem to break anything. As far as the efficacy of the feature, I really do not know, but they do have a paper on the matter. I find the bandwidth reservation/limitation to be very effective. It just takes some time to adjust such that you find the right balance where calls are preserved at the expense of others but not to the extent that the others are crippled. If that becomes the case......then more bandwidth is needed.