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I want to setup a wireless network with 3 deskops and atleast 2 laptot

I would like to implement ip multicast,voice and ip telephony,can an intergrted linksys router perform this with a wireless adapter (NIC) and if so I have never used a linksys router do i user the console cble (CLI) to configure it,
plis advice me.
rkisagara@mak.ac.ug

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Linksys has a slim Web server built into their routers and a nice little Web based configuration management interface. You can configure your router to pass or filter IP multicast. Most of the Linksys routers have a filter configuration page where you simply check to filter/pass multicast traffic. As for the VoIP/Teleplhony trafic, you should select one of the routers that allows Quality of Service (QoS) - only some routers support this. The QoS router may recognize the voice traffic automatically but at most you would have to specify the applicaation/ports to give priority to. Again, all is done easily via the Web based management utiility - no CLI needed.

Roger

Posted on Aug 08, 2008

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http://www.thunder-link.com/HUAWEI-MA5800-X7_p928.html

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Linksys adapter


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.
  • 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.
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  • End-to-End Delay
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