Need help optimizing my internet router(s) to favor VOIP devices
January 27, 2017 11:30 AM   Subscribe

My main question is how to configure my TP link router(s?) to favor the VOIP devices when I am on a call. I am having major issues on most calls. Does anyone here have experience with TP routers? Instead of QoS they have what's called "bandwidth control," where you input IP addresses and assign min/max ingress and egress values. I'm obviously not too sure how to get the most of this feature. Thanks for the help again Metafilter!

Additional information:

I have a slow internet connection: 24ms Ping / 2.32 mbps ingress / .340 mbps egress

My setup is as follows:

Main Router is TP-LINK WR841HP, which is where main internet signal connects. Connected to this are two additional TP Link routers that act as extensions, many meters away. The VOIP devices are connected to these additional routers.
There are normally 3-5 other people using the same signal on different devices, often streaming.
posted by MD_yeahright to Computers & Internet (5 answers total) 3 users marked this as a favorite
 
You can try, but unless your VoIP provider is setting QoS bits on their traffic and your ISP is respecting them you will continue to have trouble with the inbound audio no matter what you do on your side. The chance of that is very slim. Out of all the US ISPs I've used only at&t's legacy DSL product in former Southwestern Bell territory ever did, and given the speed test report I'm doubtful that's what you have.

A better option would be to forward to a landline or cell phone.

That said, if you want to try, the first step is to hope TP Link made it easy on you and automatically gives IPs with a minimum bandwidth set priority over everything else. First thing to try is setting one of your VoIP phones' IP in bandwidth control with a minimum egress bandwidth of 70kbps. If that works, and doesn't screw up anything when the phone isn't in use, set the other phones' IPs similarly.

There isn't much you can do on the inbound side since your ISP is the one deciding what to drop on its end. You can try looking in your VoIP provider's control panel to see if you can set it to only allow the g729 codec since that uses far less bandwidth (10kbps) than g711 uncompressed audio. A single dropped packet will have more effect, but the lesser number of them with audio compression generally makes it less likely to have one dropped.
posted by wierdo at 1:03 PM on January 27, 2017


Response by poster: thanks. I just did another internet speed check and it says i have 60-80% packet loss, at best its around 30%
posted by MD_yeahright at 4:11 PM on January 27, 2017


You need to fix the packet loss before worrying about QoS or anything fancier. VoIP will not work well with 30% packet loss even if it has dedicated use of the connection.
posted by primethyme at 6:04 PM on January 27, 2017 [1 favorite]


Check what the maximum transmission unit (MTU) of your router, relays, and modem are. I had such losses from excessive fragmentation that the devices could not cope with because the router was specifying a much larger size than my modem could handle.
posted by nickggully at 6:12 PM on January 27, 2017


Response by poster: thanks, how do I find MTU of devices?
posted by MD_yeahright at 7:05 PM on January 27, 2017


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