How to improve my VOIP

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Joined: 25 Feb 2012
Posts: 1

PostPosted: Sat Feb 25, 2012 15:50    Post subject: How to improve my VOIP Reply with quote
Hi everyone,
I am not able to figure this out by my own: How can I improve my voip while I am downloading through bittorrent?

Whenever I download through bittorrent on my computer while I am talking voip on my Nokia N8, the sound begins to distort and freeze for several seconds, then we can hear eachother and then it freezes again. I can sure hear a "tik tak" sound while we can't hear each other.
I thought it may have something to do with QoS settings... Here is my setup:
I have 10mb up / down speed. I set the values of QoS to 8000 uplink and downlink. Then added priorities (see attached image)

That didn't help so I can't think of anything wrong with my QoS setup, therefore I was thinking may be it has something to do with Ip filter settings:

Or just the fact that my router WHR-HP-G54 can't handle so many connections?

Any help appreciated Smile

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Ip filter settings
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Joined: 03 Jan 2010
Posts: 5630
Location: WPG, Canada

PostPosted: Tue Mar 27, 2012 5:13    Post subject: Reply with quote
8000 Kbps is 7.81 Mbps for up/down, are you use thats near what you ACTUALLY get from the ISP? With a 10/10 synchronous line usually they are fiber lines with optimal quality, at least ISPs here are..

I see several issues with your setup.. first you didn't say your build, if its not up to date as of this build, update to it. Several QoS fixes put it in a useable state finally, many CPU/RAM usage optimizations so a weak unit such as yours will enjoy that too.

Following is for after you update..

Next is your packet scheduler, HFSC is what you want for the mix of usage with your line. HFSC significantly helps regulate latency ALONG WITH the raw throughput.

Next is rtp and rtsp protocols, they should be on express. The mobile version of YouTube uses rtp/rtsp protocol and so does many other video streaming sites and software, keep them on par with httpvideo priority.

Next, sip, put to premium but not exempt, sip is not a bandwidth intensive protocol but needs good latency. Premium will give sip max priority while still taken into consideration of the max ingress/egress limits above if others are using the line.

Protocols gnutella, ftp, shoutcast may want to be added and set to bulk, skype and skypetoskype may also want to be added and set to premium.

Alternatively, as I believe your netmask priority is by priority as well and not per user speed control, is to remove ALL services and just enter the IPs using your bandwidth as bulk/standard accordingly, and your VOIP phone IP to premium, that will take ALL traffic from the phone as premium and ALL traffic from others as bulk or whatever you set them to. IF YOU TRY NETMASK YOU MUST REMOVE ALL SERVICES AS SERVICES OVERRIDE NETMASK, THEY DON'T WORK TOGETHER.

Timeouts, your TCP timeout is way too low, several sites and wikis are wrong hawking at users saying "300 seconds for 16mb RAM units 300 seconds or i kill your famery!!11", at least 900 for TCP, and UDP can go down to 90, but NO lower than 60. Max connections to 2048 and especially with the latest builds, really should handle that fine. Disable any non used features such as USB if it even has one, make sure filter multicast is checked in security settings.

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Joined: 15 Jul 2010
Posts: 376

PostPosted: Wed May 30, 2012 21:14    Post subject: Reply with quote
For what its worth,

I have had reasonable luck by subnetting my phones to a specific IP range and giving priority to those IPs, either on the same subnet for a few handsets or a complete /24 for larger deployments (30-50)

Works great on a 10Mbit Ethernet on Copper service.

My hardware is an E3000 and firmware is v15962

Hope this helps
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