Question about D-Link DGL-4100 Router (DGL4100)

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Dropping about half rx packets

I have comcast HSI and recently added digital voice. My problems began there.
I was using an old linksys befsr41v2 and it began disconnecting and having to be reset 5 or 6 times daily. I figured it was quite old and needed to be replaced.
After searching around, I decided on the dgl-4100. Its reviews showed it to be rock solid.
Once installed though, my internet speeds slowed to a crawl, like 5 minutes to load a webpage. When hooking a pc directly to the modem, no problem, speeds were fine. Comcast support said my modem signals were fine. Lan speeds between computers are fine.
So I look in the stats of the router, and over half of the wan side rx packets are being dropped.
I suspect the modem is mangling packets, but don't know how to check.

I've made new cat5 for all my runs, and installed dlink nics in all my systems (to hopefully reap some benefit from using same branded hardware).
I'm hoping you can offer me some way to check the packets between the modem and the router to see if the modem is mangling them.

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  • avikhammamet Jan 28, 2009

    I have an issue of RX packets alone being dropped in one of our suse linux servers.The drop rate is more than half.

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Proably that bad roghter not best best is linksys made by cisco ther new one n so fast 20 times of any others ok cheak online for yours see if u have a frimware or updates avb for that unit from ther site ok half duplx 10 mb sec still super fast then any network can offer u thx nick

Posted on Feb 06, 2008

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Posted on Jan 02, 2017

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Factors that Affect Voice Quality
  • Audio Compression Algorithm
    Speech signals are sampled, quantized and compressed before they are packeted and transmitted to the other end. For IP Telephony, speech signals are usually sampled at 8000 samples per second with 12-16 bits per sample. The compression algorithm plays a large role in determining the Voice Quality of the reconstructed speech signal at the other end. The SPA supports the most popular audio compression algorithms for IP Telephony: G.711 a-law and µ-law, G.726, G.729a and G.723.1. The encoder and decoder pair in a compression algorithm is known as a codec. The compression ratio of a codec is expressed in terms of the bit rate of the compressed speech. The lower the bit rate, the smaller the bandwidth required to transmit the audio packets. Voice Quality is usually lower with lower bit rate. However, Voice Quality is usually higher as the complexity of the codec gets higher at the same bit rate.
  • Silence Suppression
    The SPA applies silence suppression so that silence packets are not sent to the other end in order to conserve more transmission bandwidth. Instead, a noise level measurement can be sent periodically during silence suppressed intervals so that the other end can generate artificial comfort noise that mimics the noise at the other end using a CNG or comfort noise generator.
  • Packet Loss
    Audio packets are transported by UDP which does not guarantee the delivery of the packets. Packets may be lost or contain errors which can lead to audio sample drop-outs and distortions and lowers the perceived Voice Quality. The SPA applies an error concealment algorithm to alleviate the effect of packet loss.
  • Network Jitter
    The IP network can induce varying delay of the received packets. The RTP receiver in the SPA keeps a reserve of samples in order to absorb the Network Jitter, instead of playing out all the samples as soon as they arrive. This reserve is known as a Jitter Buffer. The bigger the Jitter Buffer, the more jitter it can absorb and the bigger the delay it can introduce. Therefore the jitter buffer size should be kept to a relatively small size whenever possible. If jitter buffer size is too small, then many late packets may be considered as lost and thus lowers the Voice Quality. The SPA can dynamically adjust the size of the jitter buffer according to the network conditions that exist during a call.
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    Certain levels of noise can be coupled into the conversational audio signals due to the hardware design. The source can be ambient noise or 60Hz noise from the power adaptor. The SPA hardware design minimizes noise coupling.
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    End-to-end delay does not affect Voice Quality directly but is an important factor in determining whether subscribers can interact normally in a conversation taking place over an IP network. Reasonable delay figure should be about 50-100ms. End-to-end delay larger than 300ms is unacceptable to most callers. The SPA supports end-to-end delays well within acceptable thresholds.

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