Call quality issue I cannot figure out

I have three Cisco SPA525G phones hooked up on the same network as my PBX, and use VoicePulse as a trunk. It works amazingly well. I have a business cable internet line via TimeWarner, at 30/5 mbps. At times when people claim they cannot hear me, nothing else is using the internet connection besides the PBX.

Recently I decided to try using an Android tablet with the same bluetooth headset I use on the SPA525G phone, as an extension of this phone system. Everyone complains they cannot hear me, it cuts in and out, when they talk I go away, it distorts, etc. I am confident this exists, because friends, family, vendors, customers, all say the same thing when I am on the tablet, and no one complains on the standard line. I doubt they are all lying as part of a conspiracy to mess with me.

I decided to record a call to my dad, to see if I could narrow this crap down. And the thing is, he says it sounds terrible, but on the recording, it sounds decent. I’d say just fine, at least for a phone call. Here is the audio clip.

I’ve tried different codecs, from g711, g722, GSM, all the same thing. SPA525G connected to the switch sounds great, tablet sounds like crap. I measured 4 mbps down and 4.5 mbps up with 89 ms latency on the tablet’s internet. I also tried using wifi on the tablet and placed it directly on top of the wireless router.

What can I do to figure out the precise cause of this problem and remedy it? Or should I toss the nexus 7 at a wall and give up on the idea of using it as an extension of my work phone system when I am not there and just get a phone?

I might get in trouble for saying this , but a couple years ago I had the misfortune to be stuck behind a TW connection, they systematically and consistently delayed my rtp traffic every few minutes for 30 seconds or so. I explicitly documented such behavior to their technical support but they denied it was happening. I stopped using them, and they were eventually and I believed successfully sued for doing that.

You can set rtp debug on and watch the traffic stream , if the timestamps and the interlacing are not close to perfect then perhaps suspect the same bad habits have returned.

Before I get in trouble for being enough of a newbie to ask how to setup rtp debug, I have a followup question for you.

Firstly; since the Cisco SPA525G wired telephones are on the same network as the PBX, which is TimeWarner, wouldn’t that cause disruption in service for them as well?

Secondly, since the audio must travel over AT&T 4G from my Nexus 7 to TimeWarner to get to the PBX to be recorded, wouldn’t this dropping show up in the recording?

Why does the recording sound fine? This is what really confuses the heck out of me. I can accept choppy audio if it exists, but I can’t hear it, and everyone else can. :frowning:

My apologies, this was three questions.

As I said I might get in trouble here, if your disrupted calls are identifiable per connection then the connection would then be identifiable by ip, so from the asterisk CLI

sip show channels

to identify the IP


rtp set debug ip n.n.n.n

as appropriate.

n.n.n.n would identify the network connection TW or AT&T or even your localhost, if there is BT involved, maybe that is the problem but only with the audio path between the channel and the BT device, then your problem is not with the carrier and the rtp stuff would show good and you would need to diagnose the BT traffic.

You didn’t mention you were trying to use 4G. It has too much latency for VoIP.

It just plain doesn’t work. If it did then everyone would be using that!

I can’t help but ask, as silly as it sounds;

On a standard cellphone, isn’t there as much latency between the phone and the cellphone tower with a phone call as there is with a 4G data plan?

I’m sorry for asking the obvious.

No 4G LTE is a DO or Data Only mode. What you are talking about is EV/DO enhanced voice and data only on the 4G infrastructure.

The voice, bearer data uses a very lossy CODEC called QCELP on a dedicated bearer path (though CDMA has 3 rake receivers so you could have 3 bearer paths, that’s how soft handoff works) designed just for the characteristics of voice.

It is not surprising that you have a clean recording, even though your voice is choppy as heard by your contact. If the source has excessive jitter, RTP packets that arrive at VoicePulse (or their upstream carrier) after they are scheduled to air are discarded, resulting in gaps in the sound. However, if no packets are lost and they arrive in order, they will all still be properly written into the recording and playback will be fine.

Don’t confuse latency with variations in latency, a.k.a. jitter. A high latency connection is lower in quality, because it’s very confusing when both parties speak at the same time. However, high latency by itself does not affect voice quality, if only one party speaks at a time.

Do a test calling 1 408 647 4636 (40-VOIP-INFO). Speak a while then press #. You’ll hear your voice played back and can judge the quality. If it’s choppy, even over Wi-Fi, test using a softphone on your computer, preferably with a wired (Ethernet) connection to your router. If your computer doesn’t have a microphone, you can use earphones as a makeshift mic. Plug them into the mic jack and speak loudly into the left earpiece.

BTW, I was unable to listen to your recording; the link opens a page which only offers the file “pbxsound.waw.exe”. Please tell me how to download the file without running a dubious executable on a Windows machine.