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Chopping Audio on VoIP

Viewing 5 posts - 1 through 5 (of 5 total)
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  • #20771

    I want to troubleshot a chopping audio problem.

    I use a Microtelco Service with an Internet LineJack hardware under a Microtelco Gateway 2.0 control (software)installed in my PC. I am using a Red Rocket or Blue Star as carriers alternatingly.

    Can anybody, bring an idea to isolate the problem?

    In advance, Tanks for everybody.


    Hi Pedro,

    I am interested in Microtelco Gateway 2.0. I was planning on signing up for their service and I was just searching the net to see if I could find any reviews before I buy.

    I am currently using Deltathree’s with a soundblaster live and a 56k dialup access. It works well most of the time but occasionally when my service provider gets too busy my connection to the internet gets slower and as a result I get choppy sound. When you speak some of your words would get cut off and when the other side speaks you hear some of the words in between. I was wondering what country are you accessing the internet from and by what means (ie 56k dialup, dsl, wireless).

    James Ballash

    Hi Guys. Using managed networks usually provide pretty good quality and a decent MOS( mean opinion score) However, depending on the routing and if the network has provided priority to VoIP calls (managed network) the “toll quality” sound may not get there. Chopping is usually due to latency and that can be enhanced with silence suppresion ON. But silence suppresion saves bandwidth. Remember depending on routing, the latency can build when more devices (routers) are in the media stream. Also I don’t know if you can ( especially with a dialup), but you may want to see if you can adjust the multiframe count to around 3. This is usually a good spot. (3 frame of voice data with 1 header)


    Can you ellaborate on how you have your VoIP system setup?

    What Codec are you using for audio compression?

    The greater the compression, the greater the likliness of “chopping” (jitter). G.711 & G.729 are usually fine, but going to G.723 or G.726 (greater compression) will increase latency leading toward jitter. As James Mentioned, Silence Suppression (VAD) can sometimes add to jitter, but I reccomend leaving it on unless you really have bandwidth to spare (haha, imagine that).

    Sherman, The fact that you are connecting over a 56k line and to the public internet is probably the main source of your problem. In order for VoIP to be very effective and have any kind of really toll quality or close to it, you need guaranteed bandwidth. When dealing with the public internet, you have no QOS in place, and you may also not getting the voice packet priority queing which is essential for VoIP to work snmoothly and clearly.

    My suggestion is to either call the carrier and discuss these topics with them to see how they handle VoIP, or try getting an ISDN line, DSL line, or Cable modem to give you more bandwidth. This should help alleve the problem, but probably not entireley elliminate it.


    is offering a free voip client, pls let me know fellas,

Viewing 5 posts - 1 through 5 (of 5 total)
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