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Audio issues - where to start troubleshooting

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UnDutchable30

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Hi UnDutchable30,

From the dropbox file please explain your issue so we can be clear.

Throughout the calling part and speech I can hear jitter, however is this what you mean ? or is the issue that there is ringing > someone picks up and answers > and the ringing starts again ?

Can you also describe whether you are using SIP trunking, or a PSTN gateway (ISDN/FXO) for your incoming/outgoing calls. Regardless, here are some suggestions:

Jitter:

If it is the jitter that is the issue please confirm that you have:

a) Enough bandwidth on the WAN link servicing this site and/or QoS techniques setup to prioritize voice: https://www.3cx.com/blog/voip-howto/real-time-network-traffic-and-qos/

Or you could try a different codec and see if that improves things - this guide is good for both calculating bandwidth and giving you the statistics for each codec:
https://www.3cx.com/blog/docs/bandwidth-utilised-for-voip/

b) If it is the nature of the call (ringing continuing after pickup) here are some posts that may help:

https://www.3cx.com/community/threa...ll-is-answered-gigaset-n510-3cx-client.57472/

https://www.3cx.com/community/threads/phone-keep-ringing.42903/
 
Also ....

Cloud hosted or on-premise?
Remote extension, or local extension?
SBC or STUN if remote?
Firewall check passed?
SIP ALG disabled?
What version of 3CX are you running?
 
One phone, or all?
From one location, or any location?
extension to extension, or outside calls?
 
Thanks for the responses and the extra questions. Let me provide some more information.

1. The error is the the "brbrbr" on top of the dialtone and the unclarity in the audio you hear, so no clear voice. The recording was done when I made an outbound call and was waiting in an IVR menu of a supplier.
2. It happens from other phones as well

-The PBX is Cloud hosted at OVH in Frankfurt with remote extensions and Stun with SIP-trunks
-Obviously Firewall check is passed, SIP is disabled and I have the latest version 15.5 installed
-The internet bandwidth is 200 Mbps down and 30 Mbps up, no WAN issues
-The router has priority routing of 70% for VOIP traffic
-Extension to extension has no problem
-Codec priority on SIP-trunk as stated by SIP provider Weepee
G711 A-law
G711 U-law
G722
G729
GSM-FR

I have also tried it via the 3CX softclient on my smartphone via 4G and the problem is present as well. So from the above I think I can limit the office network as a cause.
 
The fact that it doesn't happen on extension to extension calls could mean...a provider issue (Codec/bandwidth?), or, is there any transcoding happening in 3CX (check the Activity Log). If the sets are using one Codec and the provider another, then the PBX must make the two work together.
If the same Codec is used right the way through, then my next step would be to engage the VoIP provider, see if they can see any issues on a distorted call.

Wireshark on the WAN might also be of help.
 
Is there a VPN between the OVH instance and your office?
 
Is there a VPN between the OVH instance and your office?
Sorry for the late response, busy closing the year and some holidays. But no there is no VPN in use.
 
Sorry for the late response, busy closing the year and some holidays. But no there is no VPN in use.

Ok, given that there is no VPN and the phones are configured via STUN, there are a few possible things to look at:

1. Does each device have its own port for SIP and range of ports for media?
2. How many remote sites are there? Is it just one site with multiple STUN configured phones?
3. Have you made sure that the remote site(s) have turned off SIP ALG on the router(s)?
4. Have you considered using a SBC?
 
Hi Ari,

1. Each phone has an own network switch port and the media ranges are setup according the 3CX guidelines
2. There are no remote sites, just 1 site with multiple stun configured phones
3. The router/firewall has no SIP ALG
4. Reason for not using an SBC was that I didn't want on-premise hardware, servers etc. anymore for power consumption, noise, space etc.

Is there an advantage of having a local raspberry pi with SBC on it, even when I have only limited local calls and are the main office and not a remote office?
 
Last edited:
I haven't listed to the recording but if it's an audio quality issue and not packet loss then my money is on transcoding. OVH is a very low power instance so the CPU overhead of transcoding will show itself early. The fact that extension to extension is not having an issue confirms that. You said WeePee has G.711A as priority on the trunk and I'm betting your extensions have G.711U as the highest priority. Setting the trunk to G.711U as the highest priority (and I'd remove G.711A entirely) will likely solve your issue.

Also, your answer to Marari's question makes me think you didn't understand the question although it doesn't seem to be the issue here. Multiple STUN phones behind a single firewall require unique SIP ports and RTP ranges per 3CX. But if this was the issue you would wouldn't have audio quality issues you would have one-way audio or SIP signaling issues. That being said it is recommended to follow the 3CX recommendations which would either be configuring unique ranges per phone along with port forwarding at the router or via the use of a SBC.

This link covers the remote extension configuring:

https://www.3cx.com/docs/manual/configuring-ip-phones/#h.ul2fzupi6t22/

The last section pertains to your setup. Particularly note #2:

If you have multiple IP Phones on the same remote network configured with the same SIP and RTP ports, you might have an audio problem caused by the way certain routers implement NAT. In this scenario, each phone must have a different SIP Port and a range of RTP ports must be configured per phone. To do this, click on the Provisioning tab, change the SIP and RTP ports and perform a re-provision of each phone from the Phones node
 
You could also change the code priority for each extension to put G.711A first but that would be more work.
 
Thanks Cobaltit,

So to have the best experience on audio quality I best use G.711U-law, which I then of course select in the 3CX Admin portal under SIP, but also in every phone.
Q: Not every phone brand has all Codec present e.g. Yealink versus Snom. What to choose best in this situation?

STUN #2 explanation is clear, thanks.

So far I only experienced the audio quality, so it seems the SIP/RTP is not causing a problem, but I will consider a small SBC
 
The goal is to have the phone using the same codec as your SIP trunk so 3CX doesn't have to do any transcoding. By default all of your IP phones should have G.711U as the highest priority so changing the trunk to have G.711U as the first option is easiest. But you can also change the phones to use G.711A. It doesn't really matter which one you do as long as you just have the same codec on both sides.
 
Thanks, I have just verified the settings and I had followed the Codec order of Weepee which has G.711A first, but now I have changed all the codecs to G.711U first. Will see if it becomes better.

Anyhow thanks all for your feedback and support
 

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