Choppy Audio - Inbound Calls - Caller's Side

General Help
1

I am using a FreePBX distro PBX with firmware version: 3.211.63-10. I use Google Voice and chan_motif for my home telephone system. I have recently encountered a problem where if a person calls me I can hear them fine but the audio they receive is very choppy. This does not seem to happen on outbound calls. I have completed everything in the resolving audio issues document and my CPU usage is not maxed out. I feel like this is a problem with the system or chan_motif as it happens between all of my lines. If anyone has any suggestions it would be greatly appreciated. I hope I remembered to include all the needed details if not, let me know.

Thanks

2

I have AT&T Uverse (Hate it, don’t ever get it) and I have ports
UDP 16384-32767
TCP 5070
UDP 5070
TCP 5222
UDP 5222
TCP 443
UDP 443
TCP 5223
UDP 5223
opened to the PBX. The network is a 172.8.0.0/16 network.

3

Try opening up port 5060, and 5061 as well. if you still are having a problem put your pbx’s ip address in the DMZ and rule out port problems if it sill isn’t working. the next thing I would try is take your pbx and connect it directly to the modem. U-verse is fair I agree. Nothing better than cable internet for the home user.

4

also run a speed test and a ping test on a server close to you. www.speedtest.net.www.pingtest.net.

5

Opening ports isn’t going to do anything to improve audio quality. Especially port 5060 that is just used for call setup.

There is only one reason for choppy voice, packet loss. Yours is on the upstream side. This can be caused by congestion (too many applications trying to talk at the same time, bad cables, misconfigured speed and duplex values on routers and switches and of course poor ISP performance.

The last is more than likely your issue as DSL is an asymetric service and the upload speed is poor. You can try using a compressed CODEC. You can also try a provider that is more “on net” than Google voice with AT&T.

Use a tool like mtr on your FreePBX box to ping your local PSTN gateway (have a look at the RTP addresses when a call is up with the Asterisk RTP diagnostics) and MTR that address for a few hours.

If MTR is not installed by default it’s in the repository if you are using our distro.

6

Good to know about the ports. I would assume internal calls would have good audio. What are you getting w upload?

try something like this http://www.whichvoip.com/voip/speed_test/ppspeed.html