alaw & ulaw for both.
Well ulaw is North America, alaw is Everywhere Else. You shouldn’t have any need for alaw. Make sure that the trunk and the endpoints are all using ulaw.
You could very well be talking ulaw internally with all the extensions, remore or local, but then talking alaw over the PSTN trunk because that is what got choosen and now your PBX is transcoding those calls between alaw and ulaw.
Transcoding takes extra system resources and can cause more load on the system. When that happens you can start having audio issues. So again, make sure everything is 100% ulaw.
[quote=“Bradbpw, post:1, topic:54045”]
I have QoS set up on these routers to grant all my phones and PBX server with the highest priorities.
[/quote]I tried that. Massive Fail.
Configure QoS to allocate (reserve, partition, whatever word they use) bandwidth to the phones. “Priority” isn’t the answer, it’s reserved bandwidth.
This worked perfectly for me, after beating “Priority” QoS to death.
Look at the main CODECs used & their published bandwidth needs, multiply that by the number of phones you have + some (MAYBE) if you have plenty, set that as your QoS allocation.
Please do report back with results, good or bad?
HTH…
[quote=“Bradbpw, post:1, topic:54045”]
Could the network card in my PBX server be an issue?
[/quote]Good catch! I forgot to mention that I can make “bad calls” reports light up my endpoint by backing up the PBX or listening to recordings. Turns out, a predecessor set up the PBX with “Flexible” NIC (in ESXi) where it should have been VMXNet3.
More info: http://rickardnobel.se/vmxnet3-vs-e1000e-and-e1000-part-1/
I stand by my recommendation to allocate bandwidth for RTP packets at the router/firewall, but the PBX NIC needs love too.
These new routers I bought that were supposedly great for VoIP (Netgear X6 AC3200) only have priority based QoS. My old routers (Asus NT66U) Did have reserved bandwidth QoS. I think I’m going to go back to those routers.
I have been working with my trunk provider (1-Voip) on this. It appears that is the problem. If I switch my outgoing calls to use a different server the call audio is corrected. So, I think I’ve found the issue, but haven’t fully resolved it yet. Thanks everyone for the help!
Just an FYI, the company I work for has several PBXs running in a data center. We use VOIP Monitor to score the calls coming in and out of the main server and this software lets us go back and look at the calls. You may want to consider it to keep an eye on any problems. The software will send you an email if it get a low MOS score. You can find it at http://voipmonitor.org . The company was very helpful in getting it setup.
Thanks twohuck! I’ll take a look at that.
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