Intermittent audio

[timerider]

Every now and again I'm experiencing a drop in outgoing audio when called into via DID. It's difficult to replicate and my own test calls have been fine.
It's like the mic on the phone is suddenly muted.
Yesterday it was happening at the start of the calls, but sometimes happens later in the call.
It's a small setup, running on a reputable VPS, and the problem occurs even with just one ongoing call.

Default RTP range and signalling ports used, phones communicating over UDP.

Anyone else had this who could point me in the right direction?

 

[Adrian Fretwell]

Issues like these can be difficult to pin down. You have two at least two network sections, provider to Fusion/FreeSWITCH and F/FS to extension. Not to mention what may be happening the other side of your provider!

You can packet capture at F/FS and at your endpoint and then analyse/listen to the RTP streams in Wireshark. So you hear what is coming from your provider, what F/FS is sending to your endpoint and what is being received at your endpoint. I'm afraid it can be time consuming.

You can try enabling call recording to see if missing audio at the extension is recorded OK.

I have been in this situation many times, it has generally been caused by a network issue or network congestion, but on more than one occasion it has been a good old faulty handset cord issue or a dodgy mute button!

 

[glennbtn]

I have seen this many a time caused by a fault handset curly cable.

 

[timerider]

Interestingly, just since I set up a new polycom vvx600 that I've noticed this. I'll switch it for another one in the morning and see how that goes. No problems today as far as I could tell though, which is a little infuriating as I could capture packets all day and it not happen :/

Changing phone the easiest option so I'll try that first.
Thanks guys

 

[timerider]

Little update on this, I checked the recordings from the problematic calls and audio on both channels is fine.

When it started I was inclined to think maybe it was the caller, mobile networks past year have been terrible, but some of the problem calls have been from landlines.

In summary, looks like not the handset, my audio seems to reach the pbx, as does the callers audio. I'm using sipgate and have done for over a decade, without issues. Could it still be my side if recordings are clear?

 

[Adrian Fretwell]

So your outbound audio is getting from your extension to your PBX OK, the bit you can't prove is if your RTP is making it from your PBX to your provider, only they can help you diagnose that.

If you are on an ADSL circuit, you have less upload bandwidth that you have for download, so audio problems related to bandwidth contention tend to manifest themselves on the upload side first, unless of course the down load is being highly utilised with for example, a large download. It may be worth looking at your upload bandwidth utilisation. If you do look at it, do it yourself, most ISPs who provide usage statistics do not sample quickly enough to catch momentary spikes that can affect audio.

There is another further complication to consider. Making a call recording is less time (Real Time) critical than an endpoint reproducing the speech. In other words packets that the endpoint or SIP provider may discard because of too much jitter, the recording process can still use them to assemble a nice sounding recording. This is worth bearing in mind.

 

[timerider]

Just an update for now but not making as solved just in case...

Audio seems to have been working well, no complaints anyway. Don't ya just hate intermittent problems!

My guess, it was a routing issue at the trunk provider. Let's see how it goes though.