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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Re: dropping about half rx packets
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
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type 192.168.1.1 into your browser and hit enter. This will take you to your router page.
If you have never been to this page before this is a list of defualt passwords to enter your page. Linksys BEFW11S4, WRT54G: admin Linksys EtherFast Cable/DSL Ethernet routers: Administrator Linksys Comcast routers: comcast All other Linksys routers: [none] Default passwords: Linksys BEFW11S4: [none] Linksys Comcast routers: 1234 All other Linksys routers: admin
From here you can set your own wireless security mode and generate a passphrase or key to lock out intruders. Hope this helps...Good luck
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If your planning to connect the phone adapter behind a router make sure that the following ports are open/forwarded on the router; UDP ports 5060-5061, UDP ports 16384-32767 take note that 5060 and 5051 one are SIP ports and must be assigned to the IP address of that was assigned to by the router to the adapter or the IP that was manually configured on it.
I found a link that could help you check the setting on your linksys phone adapter
Yes you can reconnect your router. Whoever did that at Comcast had no business doing that and should be reported and return their equipment back to Comcast asap. It could appear on the bill.
I have a Linksys modem on my Comcast router and it works fine. I don't let them touch it because it uses T-Zones at home service for phone use and it uses a special router to manage it since it has a sim card to run the phone line.
Vonage equipment can be used as well.
I wonder if they were trying to put their phone service onto your system?
Only real trouble shooting I can suggest is power cycle the modem and router. If you have the Comcast Digital Voice modem make sure you push the reset button on the back as there is a battery in the unit and unplugging won't do anything.
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.
Echo Impedance mismatch between the telephone and the IP Telephony gateway phone port can lead to near-end echo. The SPA has a near end echo canceller with at least 8 ms tail length to compensate for impedance match. The SPA also implements an echo suppressor with comfort noise generator (CNG) so that any residual echo will not be noticeable.
Hardware Noise 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.
End-to-End Delay 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.