I am working with some VOIP projects. Recently a client of mine asked for a concept like VOIP on 802.11b. Certainly there are so many available products could do this possible. What is an issue here now, How does the DSS will support alternative compressions on VOIP when the Relative Telco is working and delivering analog signals for interaction and as a input. When the uplinking or Gateeway is to be completly packetized, how could we control the QoS for VOIP. Even with Freeware Gateway and Gatekeeper , how does we could reach the numbering and addressing issues like a fixed IP for the CPE.
This can be done today, problem is only roaming lets say u leave the zone, what happens to the call, how do u roam to the gsm, how is the billing done. If it is for internal setup no worries, get a QoS Voip box/router or wai for 802.11e to get its butt in gear
Setup your WLAN into a Static network so that you ca fix the IP of your CPE in reference to your Proxy server, “how does we could reach the numbering and addressing issues like a fixed IP for the CPE”… “how could we control the QoS for VOIP”, make a bandwidth management of your wireless and your wired, this will ensure QoS of yor Voip no mattter how many PCs or Laptops who wants to connect to your CPE base..
“I want to ask how many sessions in total can the system support? is it just a few sessions?”
Sessions=you mean number of calls?
it depends on the capacity of your server.. see the specs of your server so that you can see the maximum concurrent calls it can handle.. Many Servers can let you make many extensions, some actually are unlimited, but calling at the same time is limited to its capacity.. some can give you 4 concurrent calls, others more than..
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